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Live Sound Survival Guide Web

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This is a guide about how to get the best live sound possible

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Page 1: Live Sound Survival Guide Web
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ContentsCHAPTER 1: PA System Basics and Components

CHAPTER 2: Choosing and Using the Right Microphones· Microphone Types· Mic Placement and Usage· Optimizing the Sound· Cables and Connectors

CHAPTER 3: Balanced/Unbalanced Lines and Connectors· Impedance: No Worries!· Cables

CHAPTER 4:Speaker, Mixer, and Monitor Setup· Speaker Positioning· Speakers and Walls· Positioning the Mixer· Positioning Performers or Presenters· Positioning Monitors· Tonal Setup

CHAPTER 5: Mixer· Inputs and Outputs· Channel Strips· Mixer Buses· The Master Section· Mono vs. Stereo· The Importance of “Gain-Staging”· Mixer Meets Computer

CHAPTER 6: Signal Processing· Equalizer Types· Shelving EQ Applications· Parametric EQ Applications· Graphic EQ Applications· Reverb· Limiter· Compressor· Other Effects

CHAPTER 7: Amplifying Musical Instruments· Mic Placement for Acoustic Sources· Electronic Instruments· Electric Guitar and Bass “Direct” Setups· Singer/Songwriter Setup

CHAPTER 8: Sound Check· “Ringing Out” the System· Adjusting for Room Resonances· Checking Sound Levels

CHAPTER 9: Troubleshooting· Solving “Out of Phase” Problems· Dealing with Feedback· Minimizing Distortion· Reducing Hums and Buzzes

CHAPTER 10: Glossary

Editor-in-ChiefCraig Anderton

EditorsJohn KroghMichael Parker

Advertising ContactLaurie Blondin626-610-2434

Art Direction and LayoutPatrick Wong

PhotographyTheo JemisonJoshua MerrillCraig Anderton

Production Staff

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CHAPTER 1:

PA System Basics and Components“PA” stands for “Public Address,” and thekey word here is “public.” You want toamplify a band, presenter, auctioneer,lecturer, conference, worship service, orother sound source so that the audio canbe heard clearly by a large group ofpeople—your listening public.

The most common public address systemcomponents are:

Loudspeaker. Loudspeakers convertelectrical energy into acoustic energy—movingair that we can hear with our ears. In additionto loudspeakers that are like home hi-fispeakers on steroids, public address systemsoften include subwoofers. These arespeakers optimized to reproduce bass, asbass requires more power and differentspeaker construction than higher audiofrequencies. Speakers are built into cabinets,which can often mount on speaker stands,and include handles to simplify transportationand setup.

Power amplifier. A power amplifier receivesincoming audio signals, and increases theirpower so they can drive the speakers.Amplifier power is measured in watts. Thehigher the wattage, the louder the potentiallevels you can achieve with your system,although of course the speakers need to beable to handle the available power. In modern,compact systems like the M-Audio® GSRseries, the amplifiers are built into the samecabinet as the speakers. As a result, all threeelements—speaker, amplifier, and cabinet—areoptimized to work together efficiently. Inaddition, this kind of system is moretransportable (with fewer wiring issues)because it’s more self-contained.

Mixer. A mixer combines multiple signalsources, typically microphones andinstrument outputs, into a single, unifiedoutput that can then feed the poweramplifiers. However, not all situations requirea mixer. The M-Audio GSR10 and GSR12speakers have microphone and instrumentinputs—so for simple setups, it’s possible tofeed a mic or instrument directly into theGSR, and not require any other connections.

Cables. Although placing an amp and speakerwithin a single cabinet eliminates the need toconnect these two elements with a cable, youstill need cables to connect the mixer to theamplifier, and input signals to the mixer.Although you don’t need to buy ultra-expensiveaudiophile cables, quality cables are importantfor reliability, and it’s crucial to always carryspares as backup.

Microphones. These are the mirror image ofspeakers, as they convert acoustical energy

Fig. 1: The keyboardist is

using a boom mic stand to

go over the keyboard, while

the singer behind him is

using a straight,

vertical mic stand.

4 L I V E S O U N D S U R V I V A L G U I D E

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(such as a vocalist or instrument sound) intoelectrical energy that can feed the mixer orpower amplifier input. Microphone qualityrelates directly to overall sound quality, butfortunately, microphone prices have comedown over the years so you don’t have tospend much to get good quality. It’s alsopossible to use wireless microphones thattransmit sound as radio frequencies; areceiver picks up this signal, converts it toaudio, then feeds your mixer. Wirelessmicrophones help minimize clutter on stage,and are ideal for people who like to movearound stage while performing.

Microphone stands. If you’re not usingwireless microphones, you’ll need a place to“park” the mic. Even if a presenter likes tohand-hold the mic, at some point the mic will

need to be put somewhere. There are twomain types of mic stands: straight and boom(Fig. 1). Boom stands enable you to positionthe mic away from the stand if the stand getsinto the way of, for example, an instrumentalist.

Signal processors. These devices alter thebasic audio signal coming from a mic orinstrument to enhance the tone or quality, aswell as compensate for deficiencies in eitherthe equipment or the audio source. Forexample, if a lecturer’s voice has a thin quality,signal processors can make the voice soundfuller and deeper.

The room and audience. You may not thinkof them as part of a PA system, but as we’llsee later, they can have a major effect on theoverall sound.

PA SYSTEM COMPONENTS

Signal Processor

Microphone

Desktop Microphone Stand

Cable

Speaker

Mixer

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In the studio or on stage, the mic is thefirst link in the audio chain. We’ll look atwhich mic types work best for live use,then describe how to get the best soundfrom them.

MICROPHONE TYPESNo matter which mic you choose, two keyfactors are ruggedness and directionality (theability to pick up a particular sound sourcewithout picking up others).

There are two popular mic types for PAsystems (Fig. 2).

Dynamic microphones are physically ruggedand can handle high sound pressure soundlevels, so they’re the most common choice forPA systems. They also resist noise fromhandling, making them popular in hand-heldapplications (e.g., vocalists). The tradeoff is thatthe sound isn’t as refined as other technologies,but live, these differences are negligible.

Condenser microphones are common forrecording due to their excellent frequencyresponse and ability to respond to transients(rapid changes in level, as from percussion).They’re more fragile than dynamic mics, andmost models need a power supply—either froman internal battery, or from “phantom” powerthat can be supplied by all but the leastexpensive mixers (see Chapter 4 on Mixers).However, condenser mics designed for liveuse are getting more rugged, and with propercare, can hold up to the rigors of the road. Asthey’re sensitive to handling noise, they’reusually mounted on mic stands.

There are two common condenser mictypes. Small-diaphragm mics are more

sensitive, so they excel at reproducingtransients. Large-diaphragm mics tend togive a “warmer” sound, and are often usedfor vocals in the studio.

A third type of mic technology, the ribbonmicrophone, is seldom used live due to fragility.

MIC PLACEMENT AND USAGEAny mic needs to be placed so that it picks upsound optimally—but to do that, you need toknow some mic basics.

Mic Directionality

This is crucial for live use, as you want mics topick up specific sounds and discriminate againstother sounds (e.g., you don’t want a vocal mic topick up instrument sounds on the same stage).Different mics have different pickup patterns forspecific applications, sort of like cameras—some“see” only a small part of what we can see withour eyes, but there are also cameras with wide-

Fig. 2: The mic on the left is

the M-Audio SoundCheck,

an affordable dynamic

microphone. The one on the

right is the Nova, a

condenser mic that’s also

from M-Audio.

CHAPTER 2:

Choosing and Using the Right Microphones

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angle lenses, panorama cameras, etc. See Fig. 3for pickup pattern diagrams.

Cardioid (unidirectional) mics pick up only thosesounds in front of the mic. Thus the mic can“aim” at a vocalist or instrumentalist. Popularexamples of M-Audio microphones are theSoundCheck (a rugged, unidirectional dynamic),Nova® (a large-diaphragm condenser), andPulsar II (a small-diaphragm type designed forstudio use, but suitable for live).

Hypercardioid (also called supercardioid) micshave an even narrower sound field. They’re agood choice for vocalists fronting loud bands.

Omnidirectional mics pick up sounds from alldirections. They can be useful with conferences,where you have many people speaking but maynot have a mic for each person, and volumelevels aren’t particularly loud.

Figure 8 mics tend to be ribbon types, which asnoted, are rarely used live. However, somecondenser mics can also achieve this response.

Mic Placement Issues

The proximity effect is most pronounced withdynamic cardioid types—as you get closer tothe mic, the apparent bass response rises.Singers with good mic technique use this totheir advantage: On soft, intimate parts they’llhold the mic close to the mouth, to exaggeratethe bass and give a warmer sound. Acousticguitars can also use the proximity effect for amore bass-heavy sound. In any event, eitheruse it to your advantage or compensate for it.

The inverse square law states that the soundlevel drops dramatically the further the mic isfrom the sound source. This can be a problemwith vocalists who lack good mic technique,alternately “swallowing” the mic and thenbacking away, without changing their volumeto compensate.

The mic’s angle in relation to the sound sourceaffects tone. Generally, aiming the mic’s elementat the sound source results in the brightest

sound (be careful, though; with some mics the“entrance” is at the top, and with others, at theside—check your mic’s documentation). Havingthe mic at an angle produces a somewhatmellower, less “direct” sound.

Understanding these three issues helps youdecide how to mic particular sound sources.For example, soft sound sources require havingthe mic up close, but you may need to decreasethe bass response (see Chapter 6 on signalprocessing, as well as the section on

Fig. 3: The top polar pattern shows a cardioid response;

note how it rejects sounds from the rear. The middle

pattern shows an omnidirectional response, which picks

up sounds from all directions. The bottom pattern

shows a figure-eight response, so-called because it

picks up sounds from the front and back of the mic.

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Optimizing the Sound toward the end of thischapter) to compensate for the proximity effect.

With louder sound sources like windinstruments, you can back the mic away a bit,and there won’t be major volume variations ifthe player moves around a bit. When mikingsomething like a guitar amp, you have a lot oflatitude on mic placement—the standard is topoint it at the middle of the speaker in aspeaker cabinet, but different placements canyield different tones.

OPTIMIZING THE SOUNDNow that you’ve chosen your mics and set themup, here are several ways to optimize the sound.

Mic Switches

Mics often include switches for tailoring thesound, such as:

· On/off (mute) switch. This is handy when, for example, handing a mic from one person to another, or briefly turning off the mic if a sneeze or cough is imminent.

· Low-cut filter. These reduce low-frequencycontent. If a vocalist’s or lecturer’s “b” and “p” sounds cause a loud “popping,” or the proximity effect is a problem, turn the filter on.

· Pad switch. This reduces the mic’s sensitivity. For example, a loud guitar amp

might be so loud it overloads a mixer’s mic input. Enabling the pad switch will minimize distortion.

Reducing Feedback

Feedback occurs when a signal from the speakergets into the mic, and therefore gets re-amplified.Fixing feedback involves several elements of anyPA system: Speaker placement, mixer controlsettings, and mics. Using directional mics thatpoint away from speakers, and avoiding theproximity effect are two ways to reduce thepossibility of feedback at the mic itself. For moreinformation, see Chapter 9 on Troubleshooting.

Dealing with “Thin” Sounds

If you’re using two mics with an instrument likepiano, you can run into phasing problems ifeach mic picks up a different portion of thesound wave. This can result in a “thin,” unnaturalsound. If your mixer has a Phase switch for eachchannel, try flipping this for one of the mics inthe pair. If this solves the problem, fine.Otherwise, for more information on how to solvethis problem, refer to Chapter 9.

Reducing Wind and Breath Noise

The rush of air from sounds like “b” and “p”can produce nasty pops. Also, wind noise cancreate similar problems. You can use a covermade of acoustic foam that slips over themic’s head and reduces these sounds—or usea clip-on nylon mesh screen filter between thevocalist and the mic (Fig. 4).

Extending Microphone Life

Abused mics may continue to function, butlose sound quality. Always store mics in theirprotective cases.

CABLES AND CONNECTORSCables connect the various PA elements,providing a way for signals to get from oneplace to another. In the next chapter, we’ll coverhow to choose the right cable, as well as getthe best performance out of your cables.

Fig. 4: A pop filter prevents wind noise and

plosives from getting into a mic, and also blocks

breath moisture, which can have negative

effects on mic elements.

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Cables for unbalanced lines have twoconductors, while balanced lines havethree conductors (don’t confuse this withtypical stereo cables, which also havethree conductors; balanced lines are forcarrying a single, monophonic signal).Balanced lines can help rejectinterference from sources like fluorescentlight buzzes, hum, and even radiofrequencies from passing mobiletransmitters (e.g., taxis, CB radio)—aslong as they connect to inputs andoutputs whose circuitry is designed totake advantage of balanced lines. Thistechnology is particularly useful with low-level signals, as the interference might bealmost as strong as the signal itself.

Unbalanced lines are more common, andinclude guitar cords as well as most hi-ficables. Unbalanced cables don’t rejectinterference as well, but they’re generally lessexpensive, and work fine in applications withstrong signal levels.

The two different types of lines can use differentconnectors. Fig. 5 shows an unbalanced 1/4”phone plug; note that it has two sections, onefor each line. This is also called a TS type of plugbecause it has a “tip” and a “sleeve” section.

Fig. 6 shows a balanced 1/4” phone plug.These are called TRS (tip-ring-sleeve)connectors because they have three sections:The tip and sleeve (like a standard unbalancedphone connector) but also, a third “ring”section in between the tip and the sleeve.

Fig. 7 shows an XLR plug, as used forbalanced connections. The three pins connectto the three conductors.

As for jacks, you can’t tell whether 1/4” phonejacks are designed for balanced or unbalancedlines from the outside. However, mostequipment manufacturers will label their jacks(Fig. 8) so you know what type they are. XLRjacks (Fig. 9) are balanced; you can see threeholes for the three plug pins.

A relatively recent type of jack, thecombination jack (Fig. 10), gets its namebecause it combines XLR and 1/4” phone jackcapabilities—you can insert either type of plug(but of course, not both at the same time).With 1/4” phone connections, the combo jackcan work with balanced or unbalanced lines.

Another unbalanced connector, the phonoconnector (Fig. 11), is more common withconsumer gear. However, some PA mixersinclude RCA phono jacks for one or two inputsin case you need to interface with somethinglike a portable CD player. (Phono connectorsare also used for S/PDIF digital signals.)

Here are guidelines about which type of gearuses which type of line and connector.

Microphones. Almost all mics for PAapplications use balanced lines and XLRconnectors.

Synthesizers and other electricinstruments. Many of these devices offerbalanced line outputs that work with balancedor unbalanced lines. Unless the cable run isvery long or there are audible sources ofinterference, unbalanced lines will work fine.

Turntable preamps, portable CD players,etc. These consumer devices typically useunbalanced lines with RCA phono

CHAPTER 3:

Balanced/UnbalancedLines and Connectors

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Fig. 8: The M-Audio DMP3 preamp

has output jacks toward the left,

but looking at the jacks, you

wouldn’t know whether they’re

balanced or unbalanced. So,

they’re labeled as being balanced.

connections. Since cable runs are usuallyshort, unbalanced lines are not a problem.

Portable MP3 players. These use stereomini-jack and plug connectors, which resemble1/4” phone types but are smaller (Fig. 12).Most pro audio gear does not accept this typeof connector, so you if you want to use one,you’ll likely need an adapter that converts thestereo mini-jack output to two separate,unbalanced lines, typically terminated in 1/4”phone or phono plugs.

Electric guitar and bass. Almost all useunbalanced lines.

Mixer inputs. The mic inputs will acceptbalanced lines with XLR jacks. The line insusually accept either balanced or unbalancedlines via 1/4” jacks, but some higher-end gearuses XLR jacks for balanced line-level signals.

Powered speakers. Models like the M-AudioGSR series powered speakers accept the line-level signals coming from today’s mixers andinclude combo jacks, so they can interfacewith whatever connector your mixer uses.

IMPEDANCE: NO WORRIES!With today’s PA systems, impedance (which,to simplify greatly, represents the “friction”

Fig. 5: An unbalanced 1/4” phone plug. Note the tip,

the black insulating band, and ground. The flare

toward the base of the jack provides a stronger grip

when inserted into a jack.

Fig. 6: A balanced (TRS) 1/4” phone plug. Note there are

three sections, separated by black insulating bands—the

tip, the ring, and the ground. These are also called stereo

plugs when used to carry two independent left and right

channel signals.

Fig. 7: An XLR

plug, as used for

balanced line

connections.

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audio signals encounter at inputs and outputs,and is measured in ohms) isn’t something youneed to know much about—just follow a fewbasic guidelines.

Microphones. Almost all PA microphones havea low-impedance output and generate low-levelsignals. So, connect low-impedance mic outputsto low-impedance mixer mic inputs (Chapter 4)as these inputs are designed specifically toaccept low-level, low-impedance signals.

Synthesizers, turntable preamps,portable music players, etc. Theseinvariably have low-impedance inputs andgenerate high-level (“line-level”) signals.Impedance matching is not an issue if yousimply plug their outputs into mixer inputs thathandle line levels.

Electric guitar and bass. These are specialcases, because the output impedances are toohigh for mic inputs, while the signal levels aretoo high for mic inputs but too low for line-levelinputs. Some mixers have special guitar inputs;otherwise, if you plan to plug a guitar into a PA

system directly, you’ll probably need to use apreamp to match impedance and boost thelevel. However, if the guitar goes throughvarious effects, a multi-effects processor, or apedalboard, these will almost certainly generateenough level to feed line-level mixer inputs.

Powered speakers. Models like the M-AudioGSR series aren’t very sensitive to impedance,so simply plugging the mixer output into thespeaker’s input will do the job. With older, non-powered speakers, the rules regardingimpedance are pretty simple too: Match thespeaker impedance with the same impedanceoutput on your powered mixer or power amp.For example, if an amp has 4- and 8-ohmoutputs, and the speaker is rated at 8 ohms,plug it into the 8-ohm output.

CABLESHere are tips on the care and handling of cables.

· Assess what kind of cables you need to connect your various pieces of gear. Don’t buy longer cables just because they’ll work

Fig. 10: The combination XLR jack is toward the left, and accepts

both XLR and 1/4” phone plugs.

Fig. 9: The XLR jack is the large, round

connector toward the left. Also note the red LED

to its lower left, which indicates that phantom

power to the mic is enabled.

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for shorter runs too—have a selection of short and long cables, then use the appropriate length.

· Have plenty of spare cables as backup. Cables are generally reliable, but can fail (and always seem to fail at the most inopportune time).

· Use balanced lines with mics.· For relatively short cable runs—under 20-25

feet or so—unbalanced lines are fine for outputs from musical instruments like keyboards, direct outputs from guitar and bass amps, and the like. But, there’s certainlyno harm in using balanced lines if the gear accommodates them.

· Always unplug cables by grasping the plug, never by pulling on the wire.

· Avoid sharp cable bends, stepping on cables,rolling heavy objects over them, and running them near power lines or heat sources.

· Where cables are exposed to the public, tape them to the floor with non-residue gaffer’s tape. Make sure people can’t trip over them.

· Carry cable ties (available at office supply stores) to bundle cables together for a neater setup. However, don’t bundle cables carrying strong signals (like speaker connections) with cables carrying low-level signals (like mics), as there may be crosstalk.

· The metal in plugs and jacks can oxidize over time, degrading the connection. Squirt some contact cleaner (available at stores likeRadio Shack, or a more pro product like Caig’s DeoxIT) on the plug, then plug and unplug a couple dozen times. This will remove the oxidation and improve the contact.

· Connect all your cables before turning on power to your PA system. If you must connect or disconnect a cable while power is on, turn off the amplifier or powered speakers, then turn them back on again when all connections are made.

· Always have some adapters on hand, like XLR to 1/4”. You never know when to expect the unexpected.

Fig. 12: This stereo mini-plug

looks like a smaller version of a

1/4” stereo or balanced plug—and

that’s exactly what it is.

Fig. 11: The M-Audio Fast Track Pro audio interface offers balanced TRS 1/4” phone jack outputs, but note the

four RCA phono output connectors to the left of the TRS output jacks. The inset shows an RCA phono plug.

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CHAPTER 4:

Speaker, Mixer, and Monitor SetupWhere you put the speakers, mixer, mics,and performers has a huge impact on theaudience experience as well as theeffectiveness of the PA system itself. We’lllook at the optimum placement for theseelements, but first, let’s consider theproper order for turning on pieces of gear.

The most important point to remember is toturn on the amplifier (or powered speakersystems) last, after all connections havebeen made. Plugging and unpluggingdevices while the system is on can makeloud pops that are bad for your speakers andbad for your ears. Likewise, always power-upyour system with the mixer’s master volumecontrol (or powered speaker level controls)at zero—full off. Turn up the gain only whenyou know that all is well.

SPEAKER POSITIONINGThe primary rule for speaker positioning is toplace (from the audience’s viewpoint) anyspeakers in front of the mics used byperformers or presenters, with as muchdistance as possible between the mics andspeakers. This allows for the maximum gainbefore feedback. One exception is DJ setups ifa microphone is not a major part of theperformance. In this case, speakers can be on ahorizontal line with the DJ or even slightlybehind, and if the DJ has a wireless mic, he canstep behind the speaker for announcements.

When mounted on speaker mounting tripods,speakers should be higher than the audienceand tightened down so they cannot move orrotate on the tripods. If someone walks in frontof the speaker, the speaker itself should behigher than the person; in some venues,

mounting the speakers even higher providesbetter coverage.

The GSR10 and GSR12 speakers allow forstacking up to two speakers vertically, using aninterlock system that fastens the top of onespeaker to the bottom of an identical speakermodel. The extra height can improve coveragedramatically. To avoid tip-overs, make sure thecombination is stable and out the way of theaudience and performers.

To provide the best coverage with twospeakers, mount them toward the left and rightsides of the stage, and angle them slightly

14 L I V E S O U N D S U R V I V A L G U I D E

Fig. 13: In a small auditorium, angling the speakers

slightly inward toward the audience can improve

coverage. You may even be able to get away with using

only one speaker.

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toward the middle—draw an imaginary linedown the middle of the audience, and aim thespeakers to a point about three-quarters of theway toward the far end of the room (Fig. 13).Note that high frequencies emanating from aloudspeaker are highly directional, whereaslow frequencies can bend around objects in aroom. If everyone in the audience can see thespeaker(s), they’ll likely be able to hear thehigh frequencies. This also means thatsubwoofers (like the GSR18) do not need tobe raised, and in fact, will sound better if theysit on the floor or the stage (which is fortunate,as subwoofers are heavy). Furthermore,special pole mounts are available for mountinga GSR10 or GSR12 above the GSR18,providing an optimized high-frequency/low-frequency speaker pair.

SPEAKERS AND WALLSThe audience doesn’t hear only the sound fromthe speakers, but sounds reflected off thewalls and other objects. If there’s a significanttime difference between these two soundsources, music becomes less distinct, andpresenters less intelligible.

Sound travels at about one foot per millisecond(a thousandth of a second, or ms). While delaysunder about 30ms are generally not perceivedas objectionable, delays over 30ms produce anecho effect. Ideally, for an audience member thedifference in distance between the speaker andthe nearest reflective hard surface (which willproduce the loudest reflection) will be 30 feetor less. Of course this is not always possible,but the closer you can come to this ideal, thebetter. Also note that the longer the reflectionpath, the weaker the sound. So even though areflection might be more than 30 feet away, if ithas to travel a considerable distance (especiallyif there are absorptive surfaces in the room, likepeople, clothing, drapes, etc.) it won’t have asmuch influence.

Sounds bouncing off a side wall tend to be lessproblematic than sounds bouncing off a wallbehind the speaker, or in front of it. Referring toFig. 14, person “A” in the back of the hall willhear the direct sound from the speaker, and the

reflection from the back wall. This person willhear reasonably good sound quality becausethe difference between these two soundsources will be less than 30 feet, the angledsounds coming off the wall won’t be thatdifferent in length than the direct sound, and therear reflection will be much weaker anyway. Onthe other hand, person “B” will hear the soundfrom the speaker, long side reflections from thewalls, and a reflection from the back of theroom—which may be a problem if the back wallis hard and reflects a lot of the sound’s energy.This is one reason why speakers are angledtoward the middle-to-rear of the audience, asthe more direct sound will be louder than thereflections and minimize any echo effect.

POSITIONING THE MIXERAt professional concerts, the person mixing thesound is usually stationed toward the center-middle of the hall, in order to hear what theaudience hears. This is not always practical inother situations, especially since long cableruns (called “snakes”) are necessary toconnect the mixer to the mics on stage, as wellas to the speakers at the side of the stage.

Fig. 14: Person “A” may actually hear better sound quality

than person “B,” who’s closer to the speaker, depending on

a variety of factors.

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A common approach for smaller venues (e.g.,conference rooms in convention centers) is toposition the mixer off to one side, fairly close toone of the speakers. The person doing themixing will hear the direct speaker sound; whilethis doesn’t take room reflections into account,the balance among sound sources for thosehearing the direct speaker sound will be correct(or at least, be what the mixer hears).

In situations where the person doing the mixingshould not be visible for aesthetic reasons (suchas plays), or there is no room in front of thestage for a mixer, one option is to have the mixeroff to the side of the stage, just behind the edgeof a curtain. The mixer will need to rely onheadphones and during sound check, confirmthat what’s being heard in the headphones isrepresentative of what the audience is hearing.

POSITIONING PERFORMERS ORPRESENTERSAs mentioned, mics should be behind thespeakers to minimize feedback. Whenperformers use mic stands, they can bepositioned in place. Wireless mics are more ofa problem, particularly with lead singers wholike to “work the crowd,” or motivationalspeakers who roam the stage. These peopleneed to be told beforehand to avoid moving infront of the speakers, and whoever is mixingshould watch them closely and reduce thegain if it appears that their movement mightresult in feedback. Hypercardioid mics workwell in this context, and the person shouldspeak or sing close to the mic so that thevoice’s level is much louder than any soundthat might be coming from the speakers.

POSITIONING MONITORSThe performers will not hear themselvesaccurately since they are standing behind thespeakers. For some people, like presenters,this doesn’t really matter. For singers, it can becrucial because they may not be able to stayon pitch if they can’t hear themselves.

In-ear reference earphones, which resembleearbuds, are generally considered to be anoptimum solution. Otherwise, it’s common to

put a “wedge” monitor speaker in front of aperformer, pointing up to them at a 45-degreeangle (Fig. 15). Using hypercardioid micsminimizes pickup of the sound coming fromthese speakers, and the volume levels needn’tbe high as the speakers are pointing directly atthe performer from a short distance away.Both the GSR10 and GSR12 speakers canfunction as wedge monitor speakers when laidon their sides. The GSR10 is ideal for thisapplication due to its smaller size, making itless obtrusive to the audience.

TONAL SETUPAdjusting the overall tone to match a particularvenue can improve the sound qualitydramatically. One common solution is to use anequalizer (either one built in to the mixer, oradded in the signal path between mixer andamplifier), but the M-Audio GSR10 and GSR12speakers take a simpler approach by offeringfour “tuning” modes that optimize the speakersfor specific types of applications and rooms.Normal mode is the standard listening mode,with flat response. Hi-Fi gives more presence formusic playback in an application like a schooldance or when providing background music. DJmode boosts the lows and gives more definition,thus providing a “DJ sound” even at relativelylow volumes.

Voice mode is optimized when plugging a micdirectly into the GSR10 and GSR12 forconferences, boardroom meetings, seminars,and other applications where all you reallyneed is an amplified voice (no music) andaren’t using a mixer.

Fig. 15: If leaned on their sides,

the GSR10 and GSR12 can

serve as floor wedge monitors.

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CHAPTER 5:

MixersThe mixer is your PA’s traffic director—signals enter from various sources, andare routed to one or more sets of outputsthat feed the speakers. Mixers may lookdaunting, but they consist primarily ofmultiple, identical channels: Learn onechannel, and you’ve learned the others too.

INPUTS AND OUTPUTSThe main mixer specification is the number ofinputs and outputs. For example, an 8-in, 2-outmixer can accept eight different signalsources, and combine them into two outputs(also called buses). These two outputs canprovide stereo (left and right) channels.

You can think of the input signal path as avertical, downward flow into the mixer, and thebus (output signal path) as a horizontal flowfrom left to right out of the mixer (Fig. 16).Typical mixers have anywhere from four todozens of inputs.

CHANNEL STRIPSEach mixer channel has its own channel strip(also called an input module) for processing orrouting a signal before sending it to the output.A basic channel strip (Fig. 17) includes someor all of the following:

Input jack(s). This can be a low-impedanceXLR mic input, a line-level input for strongersignals, both types of inputs with a selectorswitch, a stereo signal input, or other, lesscommon options. Mixers often includedifferent inputs—a 16-input mixer might haveeight XLR mic inputs, and eight line-level1/4” inputs.

Insert jack. This allows patching externalgear, such as particular equalizers orcompressors (see Chapter 6) into the mixerchannel’s signal path. To save space, the

insert jack is usually a TRS (tip-ring-sleeve)type, and requires a special breakout cable tofeed a signal processor’s input and output.

Preamplifier. Intended to amplify low-levelsignals (e.g., mic outputs), the preamp willoffer large amounts of gain, as set by a gaincontrol. If the channel strip also offers a lineinput, the gain control will be bypassed, orcover a different, more appropriate gain range.Preamps help bring all external signals to thesame approximate level prior to being mixedtogether with the faders (described later).

Phantom power switch. When on, thissends +48V to an XLR mic input to power acondenser mic.

Low-cut filter. Although not always includedin mixers, this reduces low frequencies tominimize room rumble, mic “pops,” etc.Similarly, a high-cut filter reduces highfrequencies to lessen sibilance and hiss.

Fig. 16: Signals flow from the inputs through the

mixer channels, to the channel faders, then to the

panpots. At this point, the signals go to a master

stereo bus, which flows through the master faders,

then to the outputs.

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Clipping indicator. This isusually an LED that lights ifthe signal exceeds thepreamp’s available dynamicrange, which means youshould turn down thepreamp gain. Some inputsalso have an “activity” LEDthat lights if there’s a signalpresent (which verifies thata signal is reaching themixer channel). Anothervariation combines bothfunctions into a bi-colorLED that glows green toindicate an activity, and redto indicate overload.

Equalizer. This is a type oftone control; see Chapter 6.

Send control. See thesection on Mixer Buses.

Solo switch. Soloing achannel mutes all other inputmodules. This is useful for

hearing exactly what’s happening in a channelwithout being distracted by the other channels.

Mute switch. Muting a channel is the same

as pulling the fader all the way down. Usemute switches to keep channels out of the mixuntil right before they’re needed.

Panpot. With stereo mixers, this places theinput signal anywhere in the stereo field—fromfull left to full right.

Fader. This sets the level of the channel inrelation to the other channels. It may be alinear slider (Fig. 18), or a rotary control.

MIXER BUSESBasic mixers have two output buses. Moreadvanced models have several buses (calledsend or auxiliary buses), and you can set updifferent mixes on these different busesusing send or aux level controls. These “pickoff” part of a channel strip’s signal, and sendit to a bus in the same way a fader regulatesthe level going to the master stereo output(Fig. 19).

One common use for a mixed bus output is toset up a separate mix for singers. This isbecause if the singers can hear themselvescoming from the main speakers, then thatsound is probably getting into their mics,which encourages feedback. But setting upthe mics behind the speakers to minimizefeedback makes it difficult for the singers tohear themselves, making it tough to stay onpitch. As mentioned previously, one solution isto provide the singers with referenceearphones. A separate mix can be set upspecifically for the singer using an auxiliarybus, perhaps with vocals and melodicinstruments up loud, and drums turned down.This aux bus output feeds a separateheadphone amp, which drives the referenceearphones. Thus the singers can hearthemselves without having the monitor signalget into their mics.

Another bus application is to provide a mastervolume control for several individual channels.For example, suppose you’re mixing a churchchoir with multiple mics. You can send a signalfrom each mic to a bus, turn down the micchannel faders, then return the bus output to a

Fig. 17: Here’s a typical channel strip,from the M-Audio NRV10mixer/interface. From top to bottom,there’s an XLR mic input, a 1/4” lineinput for balanced or unbalancedlines (which you can use instead ofthe XLR in), TRS insert jack, mic/lineswitch to tailor the gain controlcharacteristics to the chosen input,preamp gain control, three EQcontrols for high/mid/lowfrequencies, two aux send controls(one to a monitor out, one to theonboard digital effects), pan control,and level fader. Note the red clippingindicator to the upper right of thelevel fader, along with a yellow lightto indicate when a channel is muted,as chosen by the Mute switch at thebottom of the channel strip.

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mixer input or send (aux) return input (Fig. 20)to blend it back in with the main output. Usethe send controls to balance each micperfectly, then if you want to raise or lower theentire choir, use the mixer input level or sendreturn control rather than adjusting each sendcontrol individually.

Send controls usually have a pre/post switch topick up the pre-channel fader signal (so the levelremains constant regardless of the fader setting)or post-channel fader, so that pulling down thefader pulls down the send to the bus as well.

Each mixer has its own way of sending signalsto buses. If space is tight and there are lots ofbuses, a channel might have a send controlalong with switches that route the send controlto a particular bus (or buses).

THE MASTER SECTIONA mixer’s master section includes “global”controls, such as:

Master output control. This varies themixer’s overall output level.

Send bus master output controls. Whilenot present on all mixers, these affect theoverall level of the bus they control. In theexample of the singer listening through

reference earphones, the bus mastercontrol would adjust the level going to thereference earphones.

Bus return controls. In the choir applicationmentioned above, you may not need to useup a mixer input for the bus return, as theremay be some “return” inputs specifically forthis purpose. These will be line level andgenerally not offer all the options of a typicalchannel strip, but usually include a levelcontrol. They can also be used as extrainputs in a pinch.

Output meters. Most mixers have at least anoutput meter, and high-end mixers mightinclude meters for individual channels. Atypical meter will consist of 10 or more LEDs,and use different colors to differentiate amongranges of levels (green for normal, yellow forclose to overload, and red for overload). Avoidletting the signal go into the red zone, as thiscan lead to distortion.

Headphone output. Most mixers have aheadphone jack with an associated level controlthat’s independent of the main output. Thismeans you can turn down the output so theaudience doesn’t hear anything, yet listen to thevarious channels on headphones to make sureeverything is working properly.

Fig. 18: Linear faders are very easy to adjust, particularly because you can move more than one at atime. These eight linear faders are part of the Avid 003 interface for Pro Tools recording software.

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MONO VS. STEREOIn most venues, mono is preferable so everyonehears the same sound. With stereo, some peoplewill be in the “sweet spot,” while others will hearmostly the left or right channel. To create a monomix with a stereo mixer, move all panpots to thecenter position.

THE IMPORTANCE OF “GAIN-STAGING”This is the process of setting levels properly—not too high, which can lead to distortion, andnot low, which can lead to hiss. There arethree main places to adjust levels in a mixer;here’s how to set them.

· Preamp gain. Set this for the maximum level short of distortion (i.e., just below wherethe clip indicator lights). It’s acceptable if the clip indicator lights briefly and very rarely; otherwise, reduce the gain.

· Master output. This will usually be calibrated in dB. 0 dB indicates that it is neither adding gain nor attenuating the signal. Negative values indicate attenuation, and positive values indicate gain. When starting a mix, set it at around –3 dB. Mixersgenerally have output meters that show the overall level leaving the mixer.

· Channel faders. Set these so that during musical peaks, the output meters peak at around –3 dB. If you need to increase the level a bit as the venue fills up, no problem—simply turn the master output to 0, or even alittle bit higher. If you need to go much higher than 0, turn up the GSR series speaker level controls.

MIXER MEETS COMPUTERSome mixers, like the M-Audio NRV10, includea computer interface that allows you to sendaudio into a computer as well as to yourspeakers. If the computer is equipped withrecording software, then it’s possible tosimultaneously record the live concert,conference, service, etc. Although computer-based recording is beyond the scope of thispublication, it’s important to realize that there’sthe potential to add recording to your liveperformance if you choose the correct mixer.

Fig. 19: The dark blue lines indicate signals going through achannel strip. The light blue lines show how signals comeoff the channel strip audio into knobs, which regulate theoutput going to the mixer’s monitor bus. The yellow linesindicate a second send of sends, which feed the mixer’sinternal effects (e.g., reverb and the like).

Fig. 20: The signals from several channels, for examplemultiple mics from a choir, go to Aux Send bus 1.Returning its output to a mixer input (in this case, the LineIn 7/8 channel) lets you use the Line In 7/8 fader to alterthe level of all mics together, without changing thebalance of the individual mics.

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CHAPTER 6:

Signal ProcessingThere are several ways to improve thesound of your PA system, both onindividual channels and the overalloutput, by using signal processors. Thinkof these as the “spices” that bring outthe best in the signal sources feeding thePA. But to use them effectively, you needto understand how they work.

EQUALIZER TYPESAn equalizer (EQ) is a tone control that helpsshape a sound’s tonal quality. Thin voices canbe made fuller, “muddy” sounds made clearer,and you can even use equalization to matchyour PA’s sonic character to specific venues.

As mentioned previously, you’ll usually findequalizers in mixer channel strips. Howeverthere are many different types, from simple tocomplex. Here’s a summary.

Shelving EQ

This is like the bass or treble control on atypical hi-fi system. It gets its name becauseit creates a “shelf” in the frequency response(Fig. 21). A treble (or “high”) shelvingequalizer boosts or cuts high frequencies,whereas a bass (or “low”) shelving equalizerboosts or cuts low frequencies. Even budgetmixers usually include treble and bassshelving EQ.

Parametric EQ

The response is different compared to shelvingequalizers, as it creates a peak (or notch) at aspecific frequency (Fig. 22). A parametric EQhas three controls.

· Boost/cut. Also called Gain, thisdetermines the amount of the peak (boost) or depth of the notch (cut).

· Frequency. This sets the frequency at whichthe boost or cut occurs.

· Bandwidth. Also called Q or resonance, thisspecifies the range of frequencies affected by the boost/cut control—from narrow to wide.

Quasi-Parametric EQ

This is like a standard parametric, but thebandwidth is fixed instead of variable.

Graphic EQ

This has multiple sliders—anywhere from three todozens—with each slider determining the boostor cut amount at a specific, fixed frequency (Fig.23). It gets its name because the sliders’physical positions show a rough approximationof the frequency response. Graphic equalizersare seldom part of a mixer’s channel strip, butinstead are included in the mixer’s mastercontrols to shape the overall output.

Multi-Stage EQ

Sophisticated mixers often combine multipleequalization stages. For example, a high-endmixer might have a four-stage EQ thatcombines a bass shelving EQ, treble shelvingEQ, and two parametric stages. This allows forversatile equalization. Even relatively low-costmixer channel strips may include three-stageequalizers with high and low shelving, and aparametric or quasi-parametric stage.

SHELVING EQ APPLICATIONSShelving EQs are best for broad, generalchanges. However, you need to be careful insituations involving microphones, as too muchadded bass or treble can lead to feedback.

· Audience compensation. People and clothing absorb sound, particularly high

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frequencies. So as a room fills up, the overallsound might seem somewhat duller. A slight high-frequency boost at the output can help.

· DJ applications. DJ audiences typically expect more bass than, say, a conference oracoustic music concert. Boosting the bass results in a more crowd-pleasing sound.

· Making presenters sound more authoritative. Conferences and lectures usually don’t require high volume levels, so you can get away with greater bass or trebleboosts than you could compared to, say, a concert. Assuming any mics have wind filters to minimize “popping” (see Chapter 2), and their low-pass filters are enabled, you can make the presenter sound more authoritative by boosting the bass a little on his or her mixer channel strip for a fuller, bigger sound.

· Increasing a presenter’s intelligibility.If people in the back row can’t understand what a presenter is saying, boosting treble to some extent can help intelligibility.

· Fixing a “muddy” sound. If the PA’s sound seems muddy, try reducing the bass at the output.

· Fixing a strident, or harsh, sound.Reducing treble at the output can give a warmer, less harsh timbre.

PARAMETRIC EQ APPLICATIONSParametric EQs are optimum for preciseapplications.

· Reducing hum. If there’s AC power- related hum in a channel, dial in a frequency of 60 Hz (50 Hz in Europe). Set a narrow(sharp) bandwidth, and cut the gain to reduce the hum.

· Reducing feedback. It’s a tricky to get this right, but parametric EQ can sometimeshelp reduce feedback. See Chapter 9 on Troubleshooting for more details.

· Separating “competing” sounds. For example, suppose a singer/songwriter is singing and playing guitar. Part of the guitar’sfrequency range will overlap with the vocal, thus competing with the vocal sound. A fairlybroad cut on the guitar channel in the vocalist’s range gives more space to the vocal. The exact frequency to cut depends on the singer; a bass or baritone can go below 100 Hz, while a high soprano can hit notes at around 1,200 Hz (with overtones ineither case going higher). Thus, consider cutting the guitar starting around 500 Hz, then adjust up or down as required by the singer.

GRAPHIC EQ APPLICATIONS· “Tuning” a room. A room is essentially a

filter; a space with thick drapes, carpets, and wood chairs will absorb more sounds—particularly higher frequencies—than a concrete room with metal chairs, which will be extremely bright. There will also be resonances that boost some frequencies, and reduce others. By listening to a known music source, like a favorite CD, you can adjust the graphic equalizer’s faders for the most accurate sound. This will change as you move around the room, so try for a good average setting rather than optimizing the sound for a specific “sweet spot.”

· General timbre changes. You can also use a graphic equalizer to create the type oftimbral changes described earlier for shelving EQ.

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Fig. 21: Shelving filter response. There is a high-frequencyboost, and low-frequency cut. The name “shelf” comesfrom the fact that after the response rises or falls, it levelsoff into a straight line.

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REVERBSome PA mixers include reverberation tosimulate the effect of being in a concert hall orlarge room. While this can sound impressive,most pros turn it off as the PA is already in aroom, so adding reverb usually results in a lessdistinct sound. The same is true for effects likedelay, echo, and “chorusing.” Less is more!

LIMITERA limiter is an electronic circuit that acts likea motor’s governor—it won’t let the soundexceed a certain level. If a mixer includes alimiter, it will usually be at the output(although some high-end mixers includelimiters on individual channels).

A limiter will have an indicator that shows whenlimiting occurs, which means the signal level(and therefore the dynamic range) is beingrestricted. You don’t want this to light veryoften; a limiter is more of a “safety valve.” If thislight illuminates a lot, you’ll probably need toreduce the overall output level control. If itlights a lot on channel strips, then you’ll likelyneed to reduce the amount of channel gain.

COMPRESSORA compressor also reduces dynamics butunlike a limiter, reduces peaks and amplifieslower-level signals to maintain a more constantlevel. The most important controls arethreshold and ratio. Lower thresholds amplifylow-level signals more, while higher ratiosreduce high-level peaks more. Whilecompressors can help with presenters atconferences where noise levels aren’t too high,amplifying low-level signals with musicalensembles can encourage feedback. For mostapplications, leave compression off.

OTHER EFFECTSThe above are common effects for PA systems,and are intended mainly to optimize the sound.However, some mixers (like the M-AudioNRV10) include effects that are more properlyconsidered “special effects.” While it’s doubtfulyou’d use these in a conference or worshipsituation, DJs can use special effects artistically

to create new types of sounds, and theatergroups can employ effects for tricks likemaking voices sound as if they’re coming froma different dimension. Mixers may also featurebuilt-in flanging (which imparts a “whooshing”character to the sound), tremolo (a rapid,cyclic fluctuation in level), delay (also calledecho, which produces the “shouting ‘hello’into a canyon” effect), and chorusing, whichmakes an individual instrument sound morelike an ensemble.

Fig. 22: Parametric filter response. A parametric filter canprovide a peak or a notch in the frequency response.

Fig. 23: A graphic EQ gets its name because the slidersdraw a “graph” of the frequency response.

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CHAPTER 7:

Amplifying Musical InstrumentsCompared to amplifying voice, theprocess of sending instruments througha PA system can be more complex. Thereare two main options for amplifyinginstruments:

· Mic acoustic sources, and feed the mic out into a mixer.

· Patch a direct electrical signal from electricalsound sources, such as electric guitar, synthesizer, electronic drums, and drum machines, etc. through patch cords into the PA system.

Let’s consider miking first. As explained in thechapter on microphones, the microphone typeand placement are the most important factorsin sound quality. So, let’s relate these tovarious miking situations.

MIC PLACEMENT FOR ACOUSTIC SOURCESVocals. While there are several recommended“best” ways for singers to use a mic (holding itaway, holding it close, singing across it,singing above it, singing into it, or acombination of these as needed to emphasizeor de-emphasize specific aspects of the song),every person’s voice requires a differenttechnique because every voice is different. Formany voices, condenser mics sound best; forothers, a dynamic mic gives the right sound.Also, refer to the chapter on Setup for tips onhow to avoid feedback when miking vocalists.

Acoustic guitar. It may not be necessary touse a mic, as many acoustic guitars havepickups and onboard preamps that feed theoutput directly into a mixer. If that isn’t an option,the mic of choice is generally a condenser typepointed between the sound hole and the bridge.

Pointing directly at the sound hole can give a“boomy” effect, while pointing directly at thebridge often sounds “thin.”

Harmony vocals. Mic placement tends to befurther away from harmony singers than leadsingers, especially if two or more singers aresharing a mic. This is also a situation wheresubtle angling can differentiate the backgroundvoices from the lead vocal.

Wind instruments. Use a dynamic mic tocope with the high sound pressure level, andback it off a foot or so from the instrumentbeing miked. With sax, the mic is generally afew inches above the bell, and pointingstraight at the sax.

Piano. Open up the lid and use two condensermics, one pointing at the high strings, the otherpointing toward the lower mid/bass strings.Make sure both are far enough away from thehammers to avoid picking up the “thunk” whenthey hit the strings.

Drums. There are several approaches tomiking drums, from having a forest of micscovering every drum to a more minimalistapproach. Less is often more, and manypeople mic drums successfully with just threeor four mics: a dynamic mic for the kick (eitherjust outside the head, or inside the drum), acondenser between the snare and high-hat topick them up, an overhead condenser micabove the drummer’s shoulder (pointing downtoward the toms and angled away from thecymbals), and perhaps one or two room mics.However, optimum placement depends on thedrum set, the desired prominence of differentdrum sounds in the mix, the drummer’sindividual setup, etc.

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Guitar and bass amps. Many amps nowinclude electrical direct outputs designedspecifically for feeding mixers and PA systems.If you can, use these to keep your life simple!But if that’s not possible, with a single-speakercabinet, point a dynamic mic at the center ofthe speaker, a few inches away from the grillecloth, or move the mic more toward the edgeof the speaker for a thinner, “tighter” sound. Ifthe cabinet has multiple speakers, try mikingjust one. Also remember that when miked, asmall, low-power amp can sound just as big asa stack of amps—without causing leakageproblems into other mics.

ELECTRONIC INSTRUMENTSThis category includes synthesizers, electronicdrums, stage pianos, drum machines, and thelike. These instruments typically produce line-level, unbalanced signals, and can plug directlyinto a suitable mixer channel input. They ofteninclude onboard signal processors, so they gettheir “sound” before going into the PA mixer,and require no adjustments other than level.

Some players with extensive stage setups(e.g., a keyboard player with multiplekeyboards, each with its own set of outputs)might also use a submixer. A submixer isconceptually like a standard mixer, but isusually designed to mix several line-levelinputs, eliminating the need for mic preamps,extensive EQ, and so on. Once the playerachieves the desired instrumental balance withthe submixer, then its master stereo output canbe sent to the main PA mixer.

ELECTRIC GUITAR AND BASS “DIRECT” SETUPSCarting around a big amp and lots of signalprocessors can get pretty tiresome, so manyguitarists downsize their setups to compactdigital processors. These digital processorsemulate the sounds of effects, amps, and evenparticular mikings of those amps.

A good example is the Avid® Eleven® Rack.It includes a variety of amp, effects, and micmodels, making it possible to get a completeguitar sound in a single, portable box. (In

addition to live performance applications,Eleven Rack doubles as a computer interfacefor Pro Tools LE® recording software.) Setupis easy, as you simply plug the guitar into theEleven Rack input, and plug the outputs intoyour PA system.

Another option is loading amp modeling andeffects software (such as Avid Eleven) into alaptop computer or even one of the morepowerful netbooks. This requires an audiointerface that connects to the computer’s USBport, FireWire port, or card slot (e.g., PCMCIAor ExpressCard) to provide physical audioinputs and outputs. Like Eleven Rack, theguitar plugs into the input, and the outputs goto the main PA mixer. Note that the audiointerface must have an input that’s designedspecifically to accommodate guitar.

Setups for electric bass are similar.

SINGER/SONGWRITER SETUPYou can insert a microphone or line-level inputinto a powered speaker such as the GSR10and GSR12, so having two of these speakers isideal for a simple, highly portable singer/songwriter setup. The mic goes into onespeaker’s mic input, while the output from anelectric piano, synthesizer, or preamp outputfrom an acoustic guitar can go into the otherspeaker’s line-level input. This eliminates theneed for a mixer, and each can have its leveladjusted independently. Note: This is a situationwhere the GSR tone options can make a hugedifference to the overall sound—try them all.

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CHAPTER 8:

Sound CheckNow that everything is set up, it’s time totest out the system and make sureeverything works and sounds as expected.

“RINGING OUT” THE SYSTEMThe optimum situation is to check out thesound with the performers or presenters whowill be playing through the system. However,this isn’t always possible, so you may need toset up hours beforehand when the rest of thepeople aren’t around.

Many sound engineers bring a portable CDplayer or other portable music player withmusic they know very well, like a favorite CD,and play that through the system after setup toget a rough idea of the system’s performance.You can use this music to check for severalsonic qualities.

Coverage. While the music is playing, walkaround the room. Ideally, the sound should beidentical wherever you are, but this is seldomthe case. You may need to change the speakerangle and/or height somewhat to obtain themost uniform coverage.

With the M-Audio GSR series speakers, it’spossible to daisy-chain up to three speakercabinets (Fig. 24). This provides manyadditional speaker placement options. Forexample, you may get better coverage bystacking a speaker on top of anotherspeaker, or by placing the two side by side—experiment to determine which is best, asacoustics vary considerably from room to room.

Placement is particularly important for highfrequencies. As you walk around the room,make sure that everyone has a straight line ofsight with the speakers (Fig. 25). If not, anglingmultiple speakers somewhat outward or

inward may give better coverage for a largersegment of the audience.

If the bass from the subwoofer seems weak insome spots, place the back of the subwooferagainst the wall, or even at the corner of twowalls. Any sound coming from thesubwoofer’s back bounces off the wall,reinforcing the main bass signal. Although notnecessarily a common technique, some soundengineers diffuse the bass sound more bypointing the subwoofer front toward a corner.This can increase the apparent amount ofbass, as well as cause the wall to vibrate,which creates additional bass.

Tone. As mentioned previously, the tone canchange dramatically depending on the numberof people in a room. Therefore, when “ringingout” a PA (a term pros use to describe thesound check process), you mainly need toanalyze the room’s character so that you canmake adjustments later. For example, if the roomhas really hard surfaces, the PA may sound alittle bright during sound check but if the roomfills up, then some of the highs will beabsorbed. It’s also worth trying the differenttone options in the GSR series speakers(Normal, Hi-Fi, DJ, and Voice—see Fig. 26) asyou may find that one option gives a betteroverall sound for a particular room than another.

Fig. 24: The GSR series speakers include a “Thru” jack thatcan pass the input signal from one GSR speaker to anotherGSR speaker’s input.

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If a room has a lot of soft surfaces—curtains,plush seats, a packed house, etc.—then you mayfind that the Hi-Fi setting gives a more balancedsound than the Normal setting.

Veterans will tell you experimentation is key,and it is. However, be careful. Switching fromone tone to another may sound great becauseit’s a change that sounds fresh, not necessarilybecause it’s better. When you make anychange, live with it for a while, then switchback to how it was originally. Do this a coupletimes to choose the “winner.” Then if

necessary, you may want to try yet anotherchange and A-B the new sound with thewinner of the previous A-B comparison.

ADJUSTING FOR ROOM RESONANCESAs mentioned in the Signal Processing chapter,some mixers have graphic equalizers at theoutputs. These make it easier to “tune” a room,but proper adjustment requires some experience.

When you listen back to a CD you know well,analyze whether any parts of the frequencyspectrum sound different than normal. Forexample, if the upper midrange sounds harsh,there may be a room resonance that accentsthis part of the spectrum. Try bringing down agraphic EQ slider in the 3–5 kHz range, andlisten for whether that makes a smoother overallresponse. Conversely, if there are lots of softsurfaces, then part of the midrange may soundattenuated, which can make presenters soundless distinct. In this case, a slight upper-midrange boost might solve the problem.

CHECKING SOUND LEVELSSome areas have ordinances that govern themaximum sound level for particular venues. Tomake sure you are in compliance, use asound level meter. These devices have acalibrated microphone and meter that showsthe level being produced at the locationwhere you’re checking levels. Sound levelmeters are available from electronics storeslike Radio Shack, although there are alsoiPhone and iPod Touch apps that providesound level metering.

Fig. 26: The GSR series’ Tone switch tailors the speaker’sresponse to different environments.

Fig. 25: Pole-mounting a speaker on top of the woofercan often provide better high-frequency coverage foryour audience.

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Once you learn the basics of running aPA, you’ll find the process to berelatively trouble-free. However, therewill be times when you encounterproblems, and these tips should helpget you through most of them.

SOLVING “OUT OF PHASE”PROBLEMSPhasing problems typically occur when you’reusing two mics on a single sound source (e.g.,miking a piano), and each mic picks up adifferent portion of the sound wave. Forexample, if one picks up the wave’s peak whileanother picks up the wave’s trough, thesounds will cancel to some degree (see Fig.27; note that we’re simplifying for the sake ofillustration). Although each mic might soundfine by itself, when combined the sound canseem “hollow” or “thin.” The sonic changewhen a jet flies overhead comes from phasechanges as you hear different reflections.

To check for phase issues, your mixer willprobably have a phase switch that reverses themic preamp’s phase. If flipping the phase switchon one of two mics creates abigger, “fatter” sound whenthey’re mixed together, thenleave the phase flipped. If there’sno phase switch, moving the mica few inches closer or furtheraway will often reduce theproblem. Your goal is the bestpossible sound when the twomic signals are mixed together.

DEALING WITHFEEDBACKFeedback is that annoying howlthat occurs when a signal fromthe speaker enters the mic, getsamplified, comes out of the

speaker again, hits the mic again, gets amplifiedsome more, and so on—creating what’s called afeedback loop. As mentioned in Chapter 2,there’s no “magic bullet” for fixing feedbackbecause the primary cause could be micplacement, speaker placement, mixer controlsettings, equalization, or even how a vocalistholds the mic. Here are tips on reducing thepotential for feedback.

Use directional mics; never use omni-directional mics. The mics should pointaway from the loudspeakers and toward thesound source.

Set up mics behind the main speakers.This reduces the chance of sound from thespeakers entering the mics (Fig. 28). Thedownside is that this also makes it difficult forvocalists or lecturers to hear what they soundlike through the speakers, which is why monitorspeakers or reference earphones are commonlyused in PA setups (see Chapter 4).

Avoid “swallowing” the mic. Vocalists whohold the mic very close often run a greater risk

CHAPTER 9:

TROUBLESHOOTING

Fig. 27: Sound waves emanate from the sound source, and two micsare positioned to pick up these sound waves. However, one picks upthe signal’s trough while the other picks up the signal’s peak, causingthe combined signal to cancel to some degree.

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of feedback than if the mic is an inch or twoaway. This is because the “proximity effect”increases the amount of bass, whichencourages feedback in the bass frequencies.

Try adjusting the equalization. Whenfeedback starts, it will be at a specificfrequency. Solo each channel to find out whichone is triggering the feedback. Set the EQ fora steep notch, then sweep the frequencycontrol until you find the frequency thatminimizes the feedback. While this may affectthe timbre of the sound going through thatchannel, the reduced feedback may be worthit. You may even be able to find anotherfeedback frequency on another channel,reduce that, and have even more level beforefeedback occurs. If the feedback seems tooccur in all channels, you may need to adjustEQ at the mixer output. A graphic equalizer willlikely not allow for fine enough adjustmentcompared to a parametric equalizer, but ifyou’re lucky, the feedback frequency mightcorrespond to a graphic EQ filter frequency.

MINIMIZING DISTORTIONDistortion can occur in several places in thesystem (see the section on Gain-Staging inChapter 5). First, check to see if any clippingindicators light when the distortion occurs. If

the indicator relates to a preamp, reduce thepreamp gain. If the indicator relates to the overallsystem (e.g., the clipping indicator on a poweramp lights), then you need to reduce the mastervolume or all the faders feeding the master output.Granted the system won’t be as loud, but a softersound is a better option than a distorted one.

Second, distortion may be occurring in thesound source itself. A damaged mic, forexample (e.g., it’s been dropped a couple times)may have a fuzzy, distorted sound. Distortion—intentional or unintentional—can also happen insomething like an instrument amplifier.

REDUCING HUMS AND BUZZESHums and buzzes are generally from one oftwo sources.

The first is lighting dimmers or fluorescentlights that generate interference. Mic preamps,cables, guitar pickups, and other sensitive orhigh-gain circuits can easily pick up thisinterference. It’s simple to test for if this is theproblem: Turn off the lights and see if theproblem goes away. If the problem relates todimmers, turning the dimmer all the way up ordown will usually solve the problem.Incandescent bulbs that don’t use dimmers areleast problematic.

Fig. 28: Top view of a stage setup. The main speakers face theaudience, with the performers and microphones set up behind themain speakers. Additional monitor speakers face the performers sothey can hear themselves.

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Some appliances can also generateinterference. If lights aren’t the problem, checkwhether unplugging appliances on the samecircuit reduces any buzzes. In many cases,purchasing a line filter and making it part ofyour PA setup is good insurance against thesetypes of buzzes and hums. However, any filtersneed to be rated for sufficient power to handlea PA system—the kind for home entertainmentsystems won’t be up to the task.

The second problem involves ground loops. Aground loop means there is more than oneground path available to a device. In Fig. 29,one path goes from device A to ground via theAC power cord’s ground terminal, but A alsosees a path to ground through the shieldedcable and AC ground of device B. Becauseground wires have some resistance (theelectronic equivalent of friction), there can be avoltage difference between the two groundlines, thus causing small amounts of current toflow through ground. This signal may getinduced into the hot conductor. The loop canalso act like an antenna for hum and radiofrequencies. Furthermore, many components ina circuit connect to ground. If that ground is“dirty,” the circuit might pick up the noise.Ground loops cause the most problems withhigh-gain circuits like mic preamps, as massiveamplification of even a couple millivolts ofnoise can be objectionable.

One supposed “fix” is to “lift”the AC ground by plugging athree-wire cord into a 3-to-2adapter. However, this isdefinitely not recommendedas it eliminates the safetyprotection afforded by agrounded chassis. Don’t do it.

A better solution is to plug allequipment into the samegrounded AC source, whichattaches all ground leads to asingle ground point (forexample, a barrier strip thatfeeds an AC outlet through ashort cord). However, make

sure that the AC source is not overloaded andis rated properly to handle the gear pluggedinto it.

A more expensive, but almost always effective,solution is isolating the piece of gear wherethe ground loop occurs by patching the audiothrough a 1:1 audio isolation transformer. Thisinterrupts the ground connection, but stillallows the signal to pass.

Another option—although the most expensiveone—is using an AC power isolationtransformer to power your gear. This can alsoprovide ancillary benefits: clean up noise fromthe AC line, reduce spikes and transients, andprovide performance almost equal to that of aseparate AC line.

If you have a lot of microprocessor-controlledgear and less-than-ideal AC power, addingisolation transformers can solve various AC-related problems and get rid of ground loops.If you just have a simple ground loop problem,then patching in audio isolation transformers inspecific audio lines may be all you need.

Granted, solving any hum and buzz problemsmay sound daunting. Fortunately, in mostcases you won’t have to worry about theseissues, particularly in smaller setups.

Fig. 29: When a device can take two separate paths to ground, thiscreates a ground loop. Whether it matters or not depends on severalfactors, such as the “dirtiness” of AC power, and also, whether the deviceis high-gain or not.

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CHAPTER 10:

GLOSSARYAmplifierA device with an input that receives weakelectrical signals, makes them stronger, thensends them to an output suitable for drivingloudspeakers. Note that while amplifiers areavailable as separate devices, they are oftenbuilt into speaker cabinets or mixers as amatter of convenience.

Auxiliary SendSee Send Bus.

CableWire that has connectors on both ends, and isdesigned to interconnect pieces of gear.

ClippingThe phenomenon that occurs when an inputsignal exceeds the headroom (dynamic range)that a system can handle, thus distorting thewaveform because it cannot be reproducedaccurately.

Clipping IndicatorA light, typically found on mixer channels andamplifier outputs, that indicates when asignal exceeds the maximum availableheadroom. When a clipping indicator lights,you need to turn down the level in thesystem (either at individual mixer channels, orthe master level if clipping occurs in thepower amplifier).

DistortionWhat happens when a waveform cannot bereproduced properly for one reason or another(e.g., inadequate power-handling capacity).

EqualizerA device that alters the signal level at specificfrequencies, for example, making the higherfrequencies softer to prevent a shrill or “tinny”sound, or increasing the level of lowerfrequencies to give more bass.

FaderA volume control, typically with a linear pathrather than a rotary one.

Input ChannelThe part of a mixer dedicated to a particularinput. It will typically have an input and fader toregulate the volume, but can also includeequalization, effects, meters, clippingindicators, and other elements designed toenhance the mixing process.

JackA connector for cables that receive or sendsignals among devices. “Female” jacks receivesignals from “male” plugs.

LevelAnother term for volume or loudness. Forexample, increasing a mixer channel’s levelcontrol will increase the volume of the signalgoing into that channel.

Line LevelA relatively strong signal level, such as whatemanates from electronic musical instruments,CD players, MP3 players, and the like.

Mic LevelA relatively weak signal level, specifically whatemanates from a microphone.

MicrophoneA transducer that converts acoustical energyto electrical energy.

Master VolumeThe control on a mixer that regulates theoverall level of the sum of all channels.

MixerA device that combines multiple signal sources(typically microphones and instrument outputs)and sends a combined signal into an amplifier.

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They include controls to adjust the relativelevel of each signal source in order to achievea proper audio balance. Mixers are specifiedas having a certain number of channels—e.g., a4-channel mixer can accept four different inputsources, while a 16-channel mixer can accept16 different input sources.

Monitor SpeakersSpeakers placed in front of performers to letthem hear themselves. These are not intendedto be heard by the audience, and may besmaller than the main speakers. Commonmonitor speakers are wedges, so calledbecause they sit on the floor, and projectsound upward and toward the performer.

PanpotWith stereo setups, this is a per-channel mixercontrol that determines whether a signal willbe heard mostly from the left side, right side,center, or for that matter, anywhere in thestereo field. When PA setups are set up inmono, the panpot is left at the center position.

Phantom PowerMost condenser microphones require a powersource in order to function correctly. Ratherthan have a power source inside the mic or ina separate unit that connects to the mic, aphantom power system sends power (typically+48 volts) along the microphone cable.Dynamic microphones do not need phantompower, but if enabled accidentally, will seldomsuffer damage from receiving phantom power.

Phone ConnectorA type of connector for balanced orunbalanced lines, typically found in semi-proand pro audio gear. It is also called a 1/4”phone connector because the barrel of theplug, or hole of the jack, has a 1/4” diameter.

Phono ConnectorA type of connector for unbalanced linestypically found on consumer hi-fi gear. Alsocalled an RCA connector.

Powered MixerA mixer that contains a built-in power amplifier,and therefore can drive speaker cabinets thatdon’t include built-in amplifiers directly.

Powered SpeakerA speaker cabinet/system that includes a built-in amplifier. This approach allows matching theamplifier and speaker.

Proximity EffectA phenomenon generally associated withdynamic cardioid mics where as the vocalistgets closer to the mic, the apparent bassresponse rises.

ReturnA mixer input that receives the output from asignal processor fed by an auxiliary or send bus.

Send BusA separate submix from a mixer that combinesvarious amounts of signals from differentchannels. This is used for two main purposes.One is to provide a mix separate from the mainmix that goes to the performers (either toheadphones or monitor speakers), as this mixmay need to be different from what theaudience hears (e.g., vocalists might want tohear themselves louder than otherinstruments). The other is to add signalprocessing to channels in various degrees byfeeding a signal processor. For example, youmight want to send a lot of vocal signal to areverb unit to add a sense of spaciousness,but send only a little bit of the guitar and noneof the drums. The knob that determines theamount of signal that’s sent is called a sendcontrol or aux send control.

XLR ConnectorA three-pin connector used with balancedlines and found in pro audio gear.

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NOTES

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NOTES

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NOTES

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Please visit m-audio.com for complete performance specifications as well as detailed information regarding compatibility.

© 2010 Avid Technology, Inc. All rights reserved. Product features, specifications, system requirements andavailability are subject to change without notice. All prices are USMSRP for U.S. only and are subject to change

without notice. Avid, M-Audio, Fast Track, Nova, Pro Tools, and Eleven are either trademarks or registered trademarksof Avid Technology, Inc. All other trademarks contained herein are the property of their respective owners.

NOTES

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