Tag Archives: Monitors

Zen And The Art Of Dialing Things In

Good instruments through neutral signal paths require very little “dialing in,” if any.

Please Remember:

The opinions expressed are mine only. These opinions do not necessarily reflect anybody else’s opinions. I do not own, operate, manage, or represent any band, venue, or company that I talk about, unless explicitly noted.

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Not long ago, Lazlo and The Dukes paid me a visit at my regular gig. They were coming off a spectacularly difficult show, and were pleased-as-punch to be in a room with manageable acoustics, a reasonably nice audio rig, and a guy to drive it all. We got settled-in via a piecemeal sort of approach. At one point, we got Steve on deck and ran his dobro through the system. He and I were both pretty happy within the span of about 30 seconds.

Later, Steve gushed about how I “just got it all ‘dialed up’ so fast.” Grateful for the compliment, and also wanting to be accurate about what occurred, I ensured Steve that he was playing a good instrument. I really hadn’t dialed anything in. I pushed up the faders and sends, and by golly, there was a nice-sounding dobro on the end of it all. I did a little experimenting with the channel EQ for FOH, wondering what would happen with a prominent midrange bump, but that was pretty optional.

In terms of “pop-culture Zen,” Steve had gotten dialed in without actually being dialed in.


Step 1: The Instrument Must Be Shaped Like Itself

The finest vocal mics I’ve ever had have been the ones in front of terrific singers. The very best signal chains I’ve ever had for drums have been the ones receiving signals derived from drums that sound killer. I’ve hurriedly hung cheap transducers in front of amazing guitar rigs, and those rigs have always come through nicely.

Whatever the “source” is, it must sound correct in and of itself. If the source uses a pickup system, that system must produce an output which sounds the way the instrument should sound.

That seems reasonable, right? The first rule of Tautology Club is the first rule of Tautology Club.

Especially with modern consoles that have tons of processing available, we can do a lot to patch problems – but that’s all we’re doing. Patching. Covering holes in things that weren’t meant to have holes. Gluing bits down and hoping it all stays together for the duration of the show. Does that sound like a shaky, uncomfortable proposition? It does because it is.

But, if the instrument is making the right noise in the room, by itself, with no extra help, then it can never NOT make the right noise in the room. We can do all kinds of things to overpower and wreck that noise by way of a PA system, but the instrument itself will always be right. In contrast, an instrument which sounds wrong may potentially be beaten into shape with the rest of the rig…but the source still doesn’t sound right. It’s completely dependent on the PA, and if the PA fails to do the job, then you’re just stuck.

An instrument which just plain “sounds good” will require very little (if any) dialing-in, so long as…

Step 2: The Rig Is Shaped Like Everything

Another way to put this is that the instrument must be filled with itself, yet the FOH PA and monitor rig must be emptied of themselves. In technical terms, the transfer function of the PA system’s total acoustical output should ideally be flat “from DC to dog-whistles.”

Let’s say you want to paint a picture. You know that the picture will be very specific, but you don’t know what that picture will be in advance. What color of canvas should you obtain? White, of course. The entire visible spectrum should be reflected by the canvas, with as little emphasis or de-emphasis on any frequency range. This is also the optimal case for a general-purpose audio system. It should impose as little of its own character as is reasonably possible upon the signals passing through.

At a practical level, this means taking the time to tune FOH and monitor world such that they are both “neutral.” Unhyped, that is. Exhibiting as flat a magnitude response as possible. To the extent that this is actually doable, this means that an instrument which is shaped like itself – sonically, I mean – retains that shape when passed through the system. This also means that if there IS a desire to adjust the tonality of the source, the effort necessary to obtain that adjustment is minimized. It is much easier to, say, add midrange to a signal when the basic path for that signal passes the midrange at unity gain. If the midrange is all scooped out (to make the rig sound “crisp, powerful, and aggressive”), then that scoop will have to first be neutralized before anything else can happen. It’s very possible to run out of EQ flexibility before you get your desired result.

Especially when talking about monitor world, this is why I’m a huge advocate for the rig to not sound “good” or “impressive” as much as it sounds “neutral.” If the actual sound of the band in the room is appropriate for the song arrangements, then an uncolored monitor rig will assist in getting everybody what they need without a whole lot of fuss. A monitor rig that’s had a lot of cool-sounding “boom” and “snap” added will, by nature, prioritize sources that emphasize those frequency ranges (and this at the expense of other sources). This can take a good acoustical arrangement and make it poor, or aggravate the heck out of an already not-so-good band configuration. It also tends to lead to feedback problems, because the critical midrange gets lost. Broadband gain is added to compensate, which combines with the effectively positive gain on the low and high-ends, and it all can end with screeching or rumbling as the loop spins out of control.

The ironic thing here is that the “netural” systems end up sounding much more impressive later on, when the show is a success. The rigs that sound impressive with walkup music, on the other hand, sometimes aren’t so nice for the actual show.

So – an audio-human with a rig that is acoustically shaped like nothing is in command of a system that is actually shaped like everything. Under the right circumstances, this means that a signal through the rig will be dialed in without any specific dialing-in being required.

Why Audio Humans Get So Bent Out Of Shape When A Mic Is Cupped

We hate it because it sounds bad, causes feedback, and makes our job harder.

Please Remember:

The opinions expressed are mine only. These opinions do not necessarily reflect anybody else’s opinions. I do not own, operate, manage, or represent any band, venue, or company that I talk about, unless explicitly noted.

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I recently posted something on my personal Facebook feed. That something was this:


A number of people found it funny.

When you really get down to it, though. It’s an “in joke.” The folks who get it have lived through a cupped-mic situation or two, and probably know why it’s a bad idea. For other folks, especially those who have been surprised by being chewed on by an irate sound-craftsperson, the whole thing might not make sense. Why would an audio human get so irritated about how a performer holds a mic? Why is it such a big deal? Why are jokes about mics being cupped such a perennial feature of live-sound forums?

The short answer is that cupped mics sound awful and tend to be feedback monsters. The long answer has to do with why.

The Physics Of How Looking Cool Sounds Bad

Microphones are curious creatures. It might sound counter-intuitive, but creating an omnidirectional mic (a mic that has essentially equal sensitivity at all angles around the element) is actually quite simple. Seal the element in a container that’s closed at the back and sides, and…there you go. Your mic is omni.

Making a directional mic is rather more involved. Directional mics require that the element NOT be housed in a box that’s sealed at the back and sides. Sound actually has to be able to arrive at the rear of the diaphragm, and it has to arrive at such a time that the combination of front and rear pressures causes cancellation. Getting this all to work, and work in a way that sounds decent, is a bear of a problem. It’s such a bear of a problem that you can’t even count on a microphone patent to tell you how it’s done. The details are kept secret – at least, if you’re asking a company like Shure.

But, anyway, the point is that a directional mic is directional because sound can reach the rear of the element. Close off the porting which allows this to happen, and the mic suddenly becomes much more omnidirectional than it was just moments before. Wrapping a hand around the head of the mic is a very efficient way of preventing certain sounds from reaching the back of the capsule, and thus, it’s a very quick way to cause a number of problems.


Fighting feedback meaningfully requires that mics be as directional as is practical. The more “screamin’ loud” the monitors and FOH have to get, the more important that directionality becomes. When setting up the show, an audio human inevitably finds a workable equilibrium ratio of gain to feedback. A highly directional mic has much lower gain in the non-sensitive directions than in the sensitive ones. This allows the sound tech to apply more gain in downstream stages (mic pres, monitor sends, FOH faders), as long as those devices result in output that the mic experiences in the “lower-gain detection arc.” At some point, a solution is arrived at – but that solution’s validity requires the gain of all devices to remain the same.

When a mic is cupped such that it becomes more omnidirectional, the established equilibrium is upset. The existing solution is invalidated, because the effective gain of the microphone itself suddenly increases. For instance, a microphone that had a gain of -10 dB at 2 kHz at 180 degrees (degrees from the mic’s front) might now have a gain of -3 dB at 2 kHz at 180 degrees. Although what I’m talking about is frequency specific, the overall result really is not fundamentally different from me reaching up to the mic-pre and adding 7 dB of gain.

Especially for a high-gain show, where the established equilibrium is already hovering close to disaster, cupping the mic will probably push us off the cliff.

Awful Tone

Intentionally omnidirectional mics can be made to sound very natural and uncolored. They don’t rely on resonance tricks to work, so very smooth and extended response is entirely achievable with due care.

Problems arise, however, when a mic becomes unintentionally omnidirectional. Directional mics are carefully tuned – intentionally “colored” – so that the resulting output is pleasant and useful for certain applications. The coloration can even be engineered so that the response is quite flat…as long as the mic element receives sound from the rear in the intended way. Much like the feedback problem I described earlier, the whole thing is a carefully crafted solution that requires the system parameters to remain in their predicted state.

A cupped microphone has its intended tuning disrupted. The mic system’s own resonant solution (which is now invalid), coupled with the resonant chamber formed by the hand around the mic, results in output which is band-limited and “peaky.” Low-frequency information tends to get lost, and the midrange can develop severe “honk” or “quack,” depending on how things shake out. At the high volumes associated with live shows, these narrow peaks of frequencies can range from merely annoying to downright painful. Vocal intelligibility can be wrecked like a ship that’s been dashed on the rocky shores of Maine.

An added bit of irony is that plenty of folks who cup microphones want a rich, powerful vocal sound…and what they end up with is something that resembles the tone of a dollar store clock-radio.

Reduced Output In Severe Cases

The worst-case scenario is when a mic is held so that the ports are obstructed, and the frontside path is ALSO obstructed. This occurs when the person using the mic wraps their whole hand around the grill, and then puts their thumb in the way of their mouth. Along with everything described above, the intervening thumb absorbs enough high-frequency content to make the mic noticeably quieter at frequencies helpful for intelligibility.

So the mic sounds bad, the singer can’t hear it, the whole mess is ready to feedback, the singer wants more monitor, and FOH needs more level.


I think you can see why sound techs get so riled by mic-cuppers. Holding a mic that way is fine if the whole performance is a pantomime. In other situations, though, it’s just bad.

EQ Or Off-Axis?

A case-study in fixing a monitor mix.

Please Remember:

The opinions expressed are mine only. These opinions do not necessarily reflect anybody else’s opinions. I do not own, operate, manage, or represent any band, venue, or company that I talk about, unless explicitly noted.

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I’m really interested in monitors. They contribute immensely to the success (or crushing failure) of a show, affect musicians in ways that are often inaudible to me, and tend to require a fair bit of management. I wrote a whole article on the topic of unsuckifying them. Some of the most interesting problems to solve involve monitor mixes, because those problems are a confluence of multiple factors that combine to smash your face in.

You know, like Devastator, the Decepticon super-robot formed by the Constructicons. The GREEN (and purple) super-robot. From the 1980s. It was kind of a pain to put him together, if I remember correctly.

Sorry, what were we talking about again?


So, my regular gig picked up a “rescue” show, because another venue shut down unexpectedly. A group called The StrangeHers was on deck, with Amanda in to play some fiddle. (Amanda is a fiddle player in high demand. If she’s not playing with a band, she is being recruited by that band. I expect that her thrash-metal debut will come shortly.) We were rushing around, trying to get monitor world sorted out. When we got to Amanda, she jumped in with a short, but highly astute question:

“The vocals are loud, but I can’t really make them out. They sound all muddy. Is there a problem with the EQ, or is it something else?”


Amanda’s monitor was equalized correctly. The lead vocal was equalized correctly. Well…that is…ELECTRONICALLY. The signal processing software acting as EQ was doing exactly what it should have been doing. Amanda’s problem had to do with effective EQ: The total, acoustical solution for her was incorrect.

In other words, yes, we had an EQ problem, but it wasn’t a problem that would be appropriately fixed with an equalizer.

One of the lessons that live-sound tries to teach – over and over again, with swift and brutal force – is that actually resolving an issue requires addressing whatever is truly precipitating that issue. You can “patch” things by addressing the symptoms, but you won’t have a fix until you get to the true, root cause.

What was precipitating the inappropriate, total EQ for Amanda could be boiled down to one fundamental factor: She wasn’t getting enough “direct” sound.

To start with, she was “off-axis” from all the other monitors she was hearing. Modern loudspeakers for live-sound applications do tend to have nice, tight, pattern control at higher frequencies. As the frequency of the reproduced content decreases, though, the output has more and more of a tendency to just “go everywhere.” Real directivity at low frequencies requires big “boxes,” as the wavelengths involved are quite large. Big boxes, however, are generally not what we want on deck, so we have to deal with what we’ve got. What we’ve got, then, is a reality where standing to the side of a monitor gets you very little in the way of frequency content that contributes to vocal intelligibility (roughly 1 kHz and above), and quite a lot of sound that contributes to vocal “mud.”

Another major factor was that the rest of what Amanda was hearing had been bounced off a boundary at least once. Any “intelligibility zone” material that made it to Amanda’s ears was significantly late when compared to everything else, and probably smeared badly from containing multiple reflections of itself. Compounding that was the issue of a room that contained both people and acoustical treatment. Most anything that was reflected back to the deck was probably missing a lot of high-frequency information. It had been heavily absorbed on the way out and the way back.

Figuring It Out

This is not to say that all of the above snapped instantly into my head when Amanda asked what was wrong. I had to have other clues in order to chase down a fix. Those clues were:

1) Before the show, I had put the mics through the monitors, walked up on deck, and listened to what it all sounded like. For the test, I had a very healthy send level from each vocal mic to the monitors that were directly behind that microphone. Vocal intelligibility was certainly happening at that time, and although things would definitely change as the room changed, the total acoustical solution wouldn’t become unrecognizably different.

2) Nobody else had complained. Although this is hardly the most reliable factor, it does figure in. If the vocals were a muddy mess everywhere, I’m betting that I would have gotten more agreement from the other band members. This suggested that the problem was local to Amanda, and by extension, that a global change (EQ on the vocal channel) would potentially create an incorrect solution for the other folks.

3) On the vocal channel, the send level to the other monitors was high in comparison to the send level to Amanda’s monitor. This was probably my biggest and most immediate clue. When other monitors are getting sends that are +9 dB in relation to another box, the performer is probably hearing mostly the garbled wash from everything OTHER than their own monitor. If the send level to Amanda’s wedge had been high, I might have concluded that the overall EQ for that particular wedge was wrong – although my encouraging, pre-show experience would have suggested that the horn had died at some point. (Ya cain’t fix THAT with an equalizer, pilgrim…)

So, with the clues that I had, I decided to try increasing the send level to Amanda’s monitor to match the send levels to the other monitors. Just like that, Amanda had a LOT more direct sound, everything was copacetic, and off we went.

Building A Small System

A guide to building a simple live-sound rig, from input to output.

Please Remember:

The opinions expressed are mine only. These opinions do not necessarily reflect anybody else’s opinions. I do not own, operate, manage, or represent any band, venue, or company that I talk about, unless explicitly noted.

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Every once in a while, I get a request for information on how to create or add-on to a system for live audio. I like to personalize this information, because rigs for show and event production are best built for specific applications. However, there is a point where insisting that everything be approached in a customized way becomes inefficient – a lot of the same ground gets covered repeatedly.


If you’re wondering what I think is required for a “small, basic, but still worthwhile” audio rig, read on. I’ll be including lots of links to vendor pages where you can buy various products.

(We’re going to go in order of signal flow, by the way.)

Input Transducers and Interfaces

For the smallest possible rig, take a distributed approach: With anything other than vocals, have the musician supply their own, personalized sound. Further, seek to avoid the whole issue of putting the output of that sound through the PA. Let a guitar amp make the guitar sound, let the drums just make their own noise in the room, and so on.

Having the ability to put more in the PA increases your potential control over the sound of the show, but outside factors can prevent that potential control from becoming actual control. Also, more inputs means more complexity, and thus more difficulty in system operation. There’s no hard and fast rule of what to be ready to reinforce, but I generally encourage folks – especially folks new to this whole thing – to ease themselves into the maelstrom that is live audio.

Wired Microphones

A decent mic will basically sound like the thing it’s being pointed at. For this reason, don’t agonize about getting specialized mics for everything right off the bat. A good “vocal” mic will be fine for many, many instrument applications. I recommend buying mics with tighter patterns (super or hypercardioid), as they can make the handling of high-gain situations much easier. Tight-pattern mics do require that the musicians – especially the vocalists – be able to use them appropriately. This issue should be considered as something for the musicians to figure out, because no system can fix everything (and a basic rig can fix even less of everything, if you take my meaning).

My current favorite mic is the EV ND767a. You might also consider the Audix OM2. Here’s a comparison of the two at Sweetwater.

Microphone Accessories

Each mic you buy will require an appropriate stand and XLR-Female to XLR-male cable.

Budget stands (like these from OnStage Stands) are just fine… if you can put in the effort to be nice to them. Gator Frameworks also has some promising offerings.

When it comes to cable, don’t overspend and don’t oversave. For a 20+ foot mic cable, paying more than $1.00 per foot is a huge premium for no benefit that I’ve been able to clearly observe. There are perfectly decent cables that can be had for as low as $0.40 per foot when purchased in a bundle. Going much below that, though, is likely to lead to problems.

I’ve been quite happy with cables I’ve gotten from Audiopile and Orange County Speaker. I’ve gotten some REALLY inexpensive cables from Unique Squared that were okay for a good while, but started having problems after a number of uses.

Wireless Mics

One word: Don’t.

Wireless is a pain in the donkey, with the FCC selling off all kinds of UHF spectrum to cellphone and computing companies, and the frequencies used for digital wireless becoming ever more crowded and hostile. Functionally, wireless transmission of audio is far more “fragile” than signal running on a cable, with all kinds of weird things that can happen outside your control.

But no matter what I say, you’re going to buy at least one wireless mic anyway, so…

Buy a digital system that operates in the 2.4 Ghz band if you want a chance at retaining your sanity in the short term. Specifically, look at the XD-V55 systems by Line 6. They’re very reasonably priced, have nice features like remote monitoring of mute status and battery level, and are the best wireless experience I’ve personally had to date.

You can, of course, go up from there.

Wireless Mic Accessories

Handheld wireless mics benefit greatly from having a stand available for every transmitter. Further, each receiver will need a cable to interface with the rest of the system.

Direct Boxes

A direct box is what I class as an “interface,” because it doesn’t convert acoustical events into electrical signals. A DI makes signals that are already electrical play nicely with pro-audio equipment when they might not otherwise. An aspect of this is also isolation, in that the DI creates what you might call an indirect connection between a console and a device. This can be very handy if the sending device (a guitar, keyboard, sampler, whatever) can’t tolerate phantom power, and the console has phantom applied to the signal line in question.

Direct boxes come in two main flavors, passive and active. Active DI boxes require external power of some kind, be that power from the wall, batteries, or phantom from the console. If you’re going to buy DI boxes, buy ACTIVE models. An active DI will work with almost anything, whereas some instrument pickups pair quite badly with passive boxes. You might as well buy units that work everywhere, and thus simplify your life. Expensive models from BSS, Radial, and Countryman are certainly nice, but there’s great value to be had in units from ART and Behringer, especially the “multiple modules in one box” offerings.

DI Box Accessories

Just like a microphone, each DI box will require an XLR-Female to XLR-Male cable. Remember that multi-module DI boxes require a cable for each individual module.

You don’t have to actually mount a rackable DI system, but you might want to. Sweetwater and Audiopile both sell quality rack cases in a huge variety of configurations.

Snakes/ Multicore Connections

Depending upon how you implement the mix and signal processing part of the rig, you may or may not require multicore cabling. If you want to send and receive a bunch of audio signals at a remote location, a snake really is a must. If you merely want to control the processing of signals from a remote location, you might be able to use your console in the same way you would use a traditional snakehead or stagebox.

If you do use a multicore, I suggest getting one that’s a little “overkill” in terms of the number of lines it contains. If a line fails, you’ll have a spare to patch to. “Headless or “fan-to-fan” snakes are a bit cheaper, but less convenient than multicores that terminate one end at a box. (You will always have to hunt for the specific line you want to connect to. It’s like a law of nature, or something.)

Audiopile would be my first choice for buying a snake I was really serious about. I have had good results with Seismic Audio fantail-to-fantail snakes, but my experience with their stagebox offerings has been mixed.

Mixing Consoles and Output Processing

This is where things get REALLY interesting.

Essentially, you have three major choices:

1) Use a relatively simple console where output processing is handled externally, and place all that at a “remote” location.

2) Use a more sophisticated console that encapsulates the output processing, and place the console at a “remote” location.

3) Use a more sophisticated console that encapsulates the output processing, leave the console close to the stage, and control the console from a remote location.

Option one takes up more space and requires more complicated physical patching, but the interface can be easier to understand at an intuitive level. Option two is compact and easy to physically patch, but the whole thing can be less intuitive for an inexperienced operator. Option three is like option two, with the added issue that control can depend entirely on an external device and network connection. If those fail, you may be in big trouble.

In every case, a console with some room to grow in terms of both inputs and outputs is a good idea. Don’t go overboard, though. You’ll end up spending a lot of money to no functional end. Especially if you’re new to all this, keep your monitor sends down to a maximum of four.

Option 1

The Console

Whatever you do, buy a console where the EQ has sweepable mids on the EQ and pre-fader auxiliary sends for each unique monitor mix you want to handle. A Soundcraft EPM8 is probably the minimum you should look for. I’ve been quite pleased with Yamaha’s mid-basic offerings, which have been revamped since I’ve bought one.

Simple, analog consoles “race” in a VERY tight pack, which means that there are lots of little permutations and many viable choices. Mackie, A&H, and Peavey are all worth looking at, and Behringer, while not having the shiniest reputation, has a knack for cramming lots of features into small cost. An XL2400 has what I would consider to be pretty darn flexible routing for a $650 mixer.

Console Accessories

Some consoles can be easily cased up or rackmounted, and some can’t.

You will need patch cables and/ or adapters with appropriate ends to get from your console outputs to your processing, one cable for each channel. As with other cables, high-dollar options really aren’t necessary. Stay somewhere in the $2/ foot range, and you’ll have patch connections that are long enough for some wiggle room and cheap enough that you won’t cry about ’em.

Also, get yourself a decent set of headphones for listening to the console’s solo bus. I have a pair of very-well-loved HD280s, and lots of other options exist. You want to look for closed-back, durable, “un-hyped” phones if you’re doing your own hunting.

The Processing

What you choose for output processing depends greatly on your own personal taste and comfort level. At the minimum, you should have an independent EQ for each mix. A really basic setup might be three mixes – one for the audience, and two monitor mixes for the stage. Things go up from there, of course.

Graphic equalizers, while not my personal favorite, are straightforward for most folks to operate. I generally recommend 31-band models over 15-band units, because you can focus in on a problem area without sledgehammering material that’s not making trouble. Peavey and dBX are good overall bets, but affordable graphics are similar to affordable consoles: All the players are very similar. I do like the features of the Behringer FBQ3102, but I also had one die after a year of use and a bumpy ride in the back of a truck.

If I’m going for an EQ-only solution, I can tell you that I vastly prefer a flexible parametric EQ over a graphic. In that realm, I have been extremely impressed by Behringer’s Feedback Destroyers. To be clear, my experience is that their automated feedback management is mediocre at best – but they are WICKED HANDY when you run them manually. You get a huge number of fully parametric filters at a very low price point. I’ve never had a major problem with any Feedback Destroyer I’ve owned. (Be aware that parametric EQ is more involved than graphic EQ. It’s not quite as “grab-n-go.” There are more choices that you have to make deliberately.)

If you want your processing to include dynamics, and also to let you have a graphic EQ combined with a parametric EQ, then a Behringer DEQ2496 is another killer device. Again, I haven’t found anything else exactly like it “in the wild,” although a Driverack PA2 is actually very close. DEQ2496 units do sometimes have problems with one of the internal connectors getting loose, but it’s an easy fix once you get the cover pulled off the device…and the connector freed from being glued down. (You will very definitely void your warranty if it’s still in effect, but hey, live a little.)

Processing Accessories

You may or may not be able to patch your snake returns (or other output cables) directly to your processing, so some adapters or patch cables might be necessary.

Also, you should definitely rack up any outboard processors you have for your system. It really does help to keep things neat and tidy.

Option 2

The Console

Digital consoles are a great route to take if you want to keep everything in one box. The downside, of course, is that the one box becomes a single point of failure. Then again, in any case, losing the console pretty much ends your day if you don’t have a spare.

Another factor to consider is that the processing available in digital consoles tends to be more fully featured while also being somewhat abstracted. This can make them overwhelming for new users, who simply don’t know what to make of all the options available.

If you’re going to be physically present at the console, you may as well get one that has a control-surface integrated in some way. An X32 Producer is a pretty natural choice, along with offerings from Presonus, Allen & Heath, and QSC.

Console Accessories

As was said earlier, rackmounting or casing the console might be possible. You can decide if you want to go to the expense or not.

The need to buy patch cables or adapters may still be there, depending on the configuration of your snake or other output lines.

And you’ll still want some headphones.

Option 3

The Console

You can leave the console on stage and mix remotely with the consoles detailed above, although the control surface might be a bit of a waste. Affordable digi-mixers that lack a surface are an interesting new product on the scene – just remember to factor in the cost of a remote-control laptop or tablet. Also, be aware that remote control is inherently a bit more “shaky” than being physically present at the mixer. It’s not horribly problematic, of course, but you have to have a contingency plan.

Surfaceless consoles at lower price points are available from Behringer, Soundcraft, and A&H. More expensive units also exist, of course.

Console Accessories

Surfaceless consoles are definitely rackable, and definitely should be.

Remember to buy the necessary patch cables or adapters for the outputs.

…but you can probably skip the headphones, because the consoles don’t currently stream the solo bus to a remote location (that I know of). That’s another downside of the number three option – to access the solo bus, you still have to be physically present at the console’s location.

Output Transducers (Speakers, That Is)

You’ll notice that I didn’t include an option for amps and passive speakers, and there’s a reason for that: This is supposed to be a simple system, and powered speakers are really the way to go to achieve simple.

Also in the service of simple is to keep your monitor wedges and FOH speakers interchangeable. Unless it’s completely inappropriate for your application, having loudspeakers that are all of the same model means that you can expect similar behavior from each box. If a failure occurs, you can swap one box for another and not have to think about it too much.

When buying powered loudspeakers, accept nothing less than an honest-to-goodness, biamped, fully processed unit. There are plenty of powered speakers that are single amped, with no processing outside of a passive crossover that is placed between the amplifier and the drivers. You want something more – something with an active crossover, basic corrective EQ applied at the factory, and an internal limiter. A peak SPL rating of greater than 120 dB @ 1 meter is also a good thing to look for, even with all the “fudging” that marketing departments apply to output numbers.

In terms of specific recommendations, I can say that PVXP12s have done very well for me. I can do “rock and roll” monitors with them in a small space, and I’m confident that they would perform equally well as an FOH unit. There are lots of other possibilities of course, provided by JBL, Mackie, Yamaha, and Behringer (just to name a few).

Loudspeaker Add-Ons

First, make sure you have the necessary cables to get from the console or snake outputs to the loudspeakers. Because speakers tend to be deployed in a rather spread-out fashion, it’s advisable to buy a bit more cable than you think you need. A 50 footer for each loudspeaker is probably a good start.

When it comes to stands for your FOH speakers, it’s good to get something a little nicer. Stands with locking collars and/ or piston assists can be a big help. I’ve used crank stands in the past, and they’re okay, but given a chance I’d make the upgrade to a more technologically advanced support.

The Biggest Accessories Of All

In the end, none of the fancy gear will mean much without power. You don’t have to buy really fancy power-conditioning equipment, but some rackmountable power units help reduce the need to fish around for a bunch of “free range” power taps. Of course, you should still keep a few freely-positionable power taps handy, along with several extension cords that use 14-gauge conductors (or something beefier, like 12-gauge). Powered speakers spread out all over creation have an alarming tendency to need those extension cords.

While there are other sundries and gadgets you can add on, going through the above should get you a working rig. As I said, this is a basic system. It won’t do everything for everybody all the time, but it should provide you with a decent start.

The Puddle Mountain Arc

If you have the space and technical flexibility, a semicircular stage layout can be pretty neat.

Please Remember:

The opinions expressed are mine only. These opinions do not necessarily reflect anybody else’s opinions. I do not own, operate, manage, or represent any band, venue, or company that I talk about, unless explicitly noted.

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Just last week, my regular gig hosted a show for The Puddle Mountain Ramblers. During the show advance, Amanda proposed an idea.

What if we set up the stage so that the layout was an arc, instead of a straight line?

I thought that was a pretty fine idea, so we went with it. The way it all came together was that fiddle, bass, and banjo were on the stage-right side, the drums were upstage center, and guitar plus another fiddle were on the stage-left side. The setup seemed very effective overall.


Visibility, Separation, and Such

The main reason for the setup was really to facilitate communication. PMR is a band that derives a good deal of comfort and confidence from the members being able to see what each other player is doing. Also, it’s just generally nice to be able to make eye contact with someone to let them know that it’s their turn for a solo. Setting up in an arc makes this much easier, because you can get essentially unobstructed sightlines from each player to every other player. An added benefit is that all the players are closer together on average, which reduces the difficulty of reading faces, identifying hand movements, and keeping time. (An arc is geometrically more compact than a line. In a linear configuration, the farthest that any two players can be from each other is the entire length of the line. Bend that same line into a circle or circle-segment, and the farthest that any two players can be from each other is the line length divided by pi. That’s a pretty significant “packing.”)

Another benefit of the configuration is (potentially) reduced drum bleed. In a traditional setup, an upstage drumkit is pretty much “firing” into the most sensitive part of all the vocal and instrument mics’ pickup patterns. In an arc layout, with the drums at the center, the direct sound from the kit enters any particular mic at some significant off-axis angle. This bleed reduction can also extend to other vocals and instruments, especially because the mics can easily be at angles greater than 90 degrees relative to other sources.

Of course, it’s important to note that – especially with wide-pattern mics, like SM58s and other cardioids – compacting the band may undo the “off-axis benefit” significantly. This is especially true for bleed from whatever source is commonly at the midpoint of the arc’s circumference, like a drumkit probably would be. For the best chance of bleed reduction, you need tighter-patterned transducers, like an ND767a, or Beta 58, or e845, or OM2, or [insert your favorite, selectively patterned mic here]. Even so, the folks closest to, and at the smallest angle from the drumkit should be the strongest singers in the ensemble, and their miced instruments should be the most able to compete with whatever is loud on deck.

A third “bit of nifty” that comes from an arc setup is that of reduced acoustical crosstalk from monitor wedge to monitor wedge. With all the wedges firing away from each other, instead of in parallel paths, the tendency for any one performer to hear the wedges adjacent to them is reduced. Each monitor mix therefore has more separation than it otherwise might, which can keep things “cleaner” overall. It may also reduce gain-hungry volume wars on the deck.


There are some caveats to putting a band on stage in a circle-segment.

The first thing to be aware of is that you tend to lose “down center” as a focal point. It’s not that you can’t put someone in there, but you have to realize that the person you’ve put down-center will no longer get the visibility and communication benefits of the arc. Also, a down-center wedge will probably be very audible to the performers standing up-center from that monitor, so you’ll have to take that into account.

The more isolated that monitor-mix sources become from one another, the more important it becomes that each monitor mix can be customized for individual performers. If you were on in-ears, for instance (the ultimate in isolated monitor feeds), separate mixes for each individual would be almost – if not entirely – mandatory. Increasing the mix-to-mix acoustical isolation pushes you towards that kind of situation. It’s not that shared mixes can’t be done in an arc, it’s just that folks have to be inclined to agree and cooperate.

A corollary to the above is that the show complexity actually tends to go up. More monitor mixes means more to manage, and an arc layout requires more thinking and cable management than a linear setup. You have to have time for a real soundcheck with careful tweaking of mixes. Throw-n-go really isn’t what you want to do when attempting this kind of layout, especially if you haven’t done it before.

Another factor to consider is that “backline” shouldn’t actually be in the back…unless you can afford to waste the space inside the arc. If at all possible, amps and instrument processing setups should utilize the empty space in front of everybody, and “fire” towards the performers (unless it’s absolutely necessary for the amps to combine with or replace the acoustical output of the PA).

If these considerations are factors you can manage, then an arc setup may be a pretty cool thing to try. For some bands, it can help “square the circle” of how to arrange the stage for the best sonic and logistical results, even if pulling it all off isn’t quite as easy as “pi.”

I’ll stop now.

Convergent Solutions

FOH and monitor world have to work together if you want the best results.

Please Remember:

The opinions expressed are mine only. These opinions do not necessarily reflect anybody else’s opinions. I do not own, operate, manage, or represent any band, venue, or company that I talk about, unless explicitly noted.

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In a small venue, there’s something that you know, even if you’re not conscious of knowing it:

The sound from the monitors on deck has an enormous effect on the sound that the audience hears. The reverse is also true. The sound from the FOH PA has an enormous effect on the sound that the musicians hear on stage.

I’m wiling to wager that there are shows that you’ve had where getting a mix put together seemed like a huge struggle. There are shows that you’ve had where – on the other hand – creating blends that made everybody happy occurred with little effort. One of the major factors in the ease or frustration of whole-show sound is “convergence.” When the needs of the folks on deck manage to converge with the needs of the audience, sound reinforcement gets easier. When those needs diverge, life can be quite a slog.

Incompatible Solutions

But…why would the audience’s needs and the musicians’ needs diverge?

Well, taste, for one thing.

Out front, you have an interpreter for the audience, i.e. the audio human. This person has to make choices about what the audience is going to hear, and they have to do this through the filter of their own assumptions. Yes, they can get input from the band, and yes, they will sometimes get input from the audience, but they still have to make a lot of snap decisions that are colored by their immediate perceptions.

When it comes to the sound on deck, the noise-management professional becomes more of an “executor.” The tech turns the knobs, but there can be a lot more guidance from the players. The musicians are the ones who try to get things to match their needs and tastes, and this can happen on an individual level if enough monitor mixes are available.

If the musicians’ tastes and the tech’s taste don’t line up, you’re likely to have divergent solutions. One example I can give is from quite a while ago, where a musician playing a sort of folk-rock wanted a lot of “kick” in the wedges. A LOT of kick. There was so much bass-drum material in the monitors that I had none at all out front. Even then, it was a little much. (I was actually pretty impressed at the amount of “thump” the monitor rig would deliver.) I ended up having to push the rest of the mix up around the monitor bleed, which made us just a bit louder than we really needed to be for an acoustic-rock show.

I’ve also experienced plenty of examples where we were chasing vocals and instruments around in monitor world, and I began to get the sneaky suspicion that FOH was being a hindrance. More than once, I’ve muted FOH and heard, “Yeah! That sounds good now.” (Uh oh.)

In any case, the precipitating factors differ, but the main issue remains the same: The “solutions” for the sound on stage and the sound out front are incompatible to some degree.

I say “solutions” because I really do look at live-sound as a sort of math or science “problem.” There’s an outcome that you want, and you have to work your way through a process which gets you that outcome. You identify what’s working against your desired result, find a way to counteract that issue, and then re-evaluate. Eventually, you find a solution – a mix that sounds the way you think it should.

And that’s great.

Until you have to solve for multiple solutions that don’t agree, because one solution invalidates the others.

Live Audio Is Nonlinear Math

The analogy that I think of for all this is a very parabolic one. Literally.

If you remember high school, you probably also remember something about finding “solutions” for parabolic curves. You set the function as being equal to zero, and then tried to figure out the inputs to the function that would satisfy that condition. Very often, you would get two numbers as solutions because nonlinear functions can output zero more than once.

In my mind, this is a pretty interesting metaphor for what we try to do at a show.

For the sake of brevity, let’s simplify things down so that “the sound on stage” and “the sound out front” are each a single solution. If we do that, we can look at this issue via a model which I shall dub “The Live-Sound Parabola.” The Live-Sound Parabola represents a “metaproblem” which encompasses two smaller problems. We can solve each sub-problem in isolation, but there’s a high likelihood that the metaproblem will remain unsolved. The metaproblem is that we need a good show for everyone, not just for the musicians or just for the audience.

In the worst-case scenario, neither sub-problem is even close to being solved. The show is bad for everybody. Interestingly, the indication of the “badness” of the show is the area under the curve. (Integral calculus. It’s everywhere.) In other words, the integral of The Live Sound Parabola is a measure of how much the sub-solutions functionally diverge.


(Sorry about the look of the graphs. Wolfram Alpha doesn’t give you large-size graphics unless you subscribe. It’s still a really cool website, though.)


A fairly common outcome is that we don’t quite solve the “on deck” and “out front” problems, but instead arrive at a compromise which is imperfect – but not fatally flawed. The area between the curve and the x-axis is comparatively small.


When things really go well, however, we get a convergent solution. The Live-Sound Parabola becomes equal to zero at exactly one point. Everybody gets what they want, and the divergence factor (the area under the curve) is minimized. (It’s not eliminated, but simply brought to its minimum value.)


What’s interesting is that The Live Sound Parabola still works when the graph drops below zero. When it does, it’s showing a situation where two diverging solutions actually work independently. This is possible with in-ear monitors, where the solution for the musicians can be almost (if not completely) unaffected by the FOH mix. The integral still shows how much divergence exists, but in this case the divergence is merely instructive rather than problematic.


How To Converge

At this point, you may be wanting to shout, “Yeah, yeah, but what do we DO?”

I get that.

The first thing is to start out as close to convergence as possible. The importance of this is VERY high. It’s one of the reasons why I say that sounding like a band without any help from sound reinforcement is critical. It’s also why I discourage audio techs from automatically trying to reinvent everything. If the band already sounds basically right, and the audio human does only what’s necessary to transfer that “already right sound” to the audience, any divergence that occurs will tend to be minimal. Small divergence problems are simple to fix, or easy to ignore. If (on the other hand) you come out of the gate with a pronounced disagreement between the stage and FOH, you’re going to be swimming against very strong current.

Beyond that, though, you need two things: Time, and willingness to use that time for iteration.

One of my favorite things to do is to have a nice, long soundcheck where the musicians can play in the actual room. This “settling in” period is ideally started with minimal PA and minimal monitors. The band is given a chance to get themselves sorted out “acoustically,” as much as is practical. As the basic onstage sound comes together, some monitor reinforcement can be added to get things “just so.” Then, some tweaks at FOH can be applied if needed.

At that point, it’s time to evaluate how much the house and on-deck solutions are diverging. If they are indeed diverging, then some changes can be applied to either or both solutions to correct the problem. The musicians then continue to settle in for a bit, and after that you can evaluate again. You can repeat this process until everybody is satisfied, or until you run out of time.

With a seasoned band and experienced audio human, this iteration can happen very fast. It’s not instant, though, which is another reason to actually budget enough time for it to happen. Sometimes that’s not an option, and you just have to “throw and go.” However, I have definitely been in situations where bands wanted to be very particular about a complex show…after they arrived with only 30 minutes left until downbeat. It’s not that I didn’t want to do everything to help them, it’s just that there wasn’t time for everything to be done. (Production craftspersons aren’t blameless, either. There are audio techs who seem to believe that all shows can be checked in the space of five minutes, and remain conspicuously absent from the venue until five minutes is all they have. Good luck with that, I guess.)


If everybody does their homework, and is willing to spend an appropriate amount of prep-time on show day, your chances of enjoying some convergent solutions are much higher.

A Vocal Addendum

Forget about all the “sexy” stuff. Get ’em loud, and let ’em bark.

Please Remember:

The opinions expressed are mine only. These opinions do not necessarily reflect anybody else’s opinions. I do not own, operate, manage, or represent any band, venue, or company that I talk about, unless explicitly noted.

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This article is a follow-on to my piece regarding the unsuckification of monitors. In a small-venue context, vocal monitoring is probably more important than any other issue for the “on deck” sound. Perhaps surprisingly, I didn’t talk directly about vocals and monitors AT ALL in the previous article.

But let’s face it. The unsuckification post was long, and meant to be generalized. Putting a specific discussion of vocal monitoring into the mix would probably have pushed the thing over the edge.

I’ll get into details below, but if you want a general statement about vocal monitors in a small-venue, “do-or-die,” floor-wedge situation, I’ll be happy to oblige: You do NOT need studio-quality vocals. You DO need intelligible, reasonably smooth vocals that can be heard above everything else. Forget the fluff – focus on the basics, and do your preparation diligently.

Too Loud Isn’t Loud Enough

One of the best things to ever come out of Pro Sound Web was this quiz on real-world monitoring. In particular, answer “C” on question 16 (“What are the main constituents of a great lead vocal mix?”) has stuck with me. Answer C reads: “The rest of the band is hiding 20 feet upstage because they can’t take it anymore.”

In my view, the more serious rendering of this is that vocal monitors should, ideally, make singing effortless. Good vocal monitors should allow a competent vocalist to deliver their performance without straining to hear themselves. To that end, an audio human doing show prep should be trying to get the vocal mics as loud as is practicable. In the ideal case, a vocal mic routed through a wedge should present no audible ringing, while also offering such a blast of sound that the singer will ask for their monitor send to be turned down.

(Indeed, one of my happiest “monitor guy” moments in recent memory occurred when a vocalist stepped up to a mic, said “Check!”, got a startled look on his face, and promptly declared that “Anyone who can’t hear these monitors is deaf.”)

Now, wait a minute. Doesn’t this conflict with the idea that too much volume and too much gain are a problem?


Vocal monitors are a cooperative effort amongst the audio human, the singer(s), and the rest of the band. The singer has to have adequate power to perform with the band. The band has to run at a reasonable volume to play nicely with the singer. If those two conditions are met (and assuming there are no insurmountable equipment or acoustical problems), getting an abundance of sound pressure from a monitor should not require a superhuman effort or troublesome levels of gain.

So – if you’re prepping for a band, dial up as much vocal volume as you can without causing a loop-gain problem. If the vocals are tearing people’s heads off, you can always turn it down. Don’t be lazy! Get up on deck and listen to what it sounds like. If there are problem areas at certain frequencies, then get on the appropriate EQ and tame them. Yes, the feedback points can change a bit when things get moved around and people get in the room, but that’s not an excuse to just sit on your hands. Do some homework now, and life will be easier later.

Don’t Squeeze Me, Bro

A sort of corollary to the above is that anything which acts to restrict your vocal monitor volume is something you should think twice about. If you were thinking about inserting a compressor in such a way that it would affect monitor world, think again.

A compressor reduces dynamic range by reducing gain on signals that exceed a preset threshold. For a vocalist, this means that the monitor level of their singing may no longer track in a 1:1 ratio with their output at the mic. They sing with more force, but the return through the monitors doesn’t get louder at the same rate. If the singer is varying their dynamics to track with the band, this failure of the monitors to stay “in ratio” can cause the vocals to become swamped.

And, in certain situations, monitors that don’t track with vocal dynamics can cause a singer to hurt themselves. They don’t hear their voice getting as loud as it should, so they push themselves harder – maybe even to the point that they blow out their voice.

Of course, you could try to compensate for the loss of level by increasing the output or “makeup” gain on the compressor, but oh! There’s that “too much loop gain” problem again. (Compressors do NOT cause feedback. That’s a myth. Steady-state gain applied to compensate for compressor-applied, variable gain reduction, on the other hand…)

The upshot?

Do NOT put a compressor across a vocalist such that monitor world will be affected. (The exception is if you have been specifically asked to do so by an artist that has had success with the compressor during a real, “live-fire” dress rehearsal.) If you don’t have an independent monitor console or monitor-only channels, then bus the vocals to a signal line that’s only directly audible in FOH, and compress that signal line.

The Bark Is The Bite

One thing I have been very guilty of in the past, and am still sometimes guilty of, is dialing up a “sounds good in the studio” vocal tone for monitor world. That doesn’t sound like it would be a problem, but it can be a huge one.

The issue at hand is that what sounds impressive in isolation often isn’t so great when the full band is blasting away. This is very similar to guitarists who have “bedroom” tone. When we’re only listening to a single source, we tend to want that source to consume the entire audible spectrum. We want that single instrument or voice to have extended lows and crisp, snappy HF information. We will sometimes dig out the midrange in order to emphasize the extreme ends of the audible spectrum. When all we’ve got to listen to is one thing, this can all sound very “sexy.”

And then the rest of the band starts up, and our super-sexy, radio-announcer vocals become the wrong thing. Without a significant amount of midrange “bark,” the parts of the spectrum truly responsible for vocal audibility get massacred by the guitars. And drums. And keyboards. All that’s left poking through is some sibilance. Then, when you get on the gas to compensate, the low-frequency material starts to feed back (because it’s loud, and the mic probably isn’t as directional as you think at low frequencies), and the high-frequency material also starts to ring (because it’s loud, and probably has some nasty peaks in it as well).

Yes – a good monitor mix means listenable vocals. You don’t want mud or nasty “clang” by any means, but you need the critical midrange zone – say, 500 Hz to 3 KHz or 4 KHz – to be at least as loud as the rest of the audible spectrum in the vocal channel. Midrange that jumps at you a little bit doesn’t sound as refined as a studio recording, but this isn’t the studio. It’s live-sound. Especially on the stage, hi-fi tone often has to give way to actually being able to differentiate the singer. There are certainly situations where studio-style vocal tone can work on deck, but those circumstances are rarely encountered with rock bands in small spaces.

Stay Dry

An important piece of vocal monitoring is intelligibility. Intelligibility has to do with getting the oh-so-important midrange in the right spot, but it also has to do with signals starting and stopping. Vocal sounds with sharply defined start and end points are easy for listeners to parse for words. As the beginnings and ends of vocal sounds get smeared together, the difficulty of parsing the language goes up.

Reverb and delay (especially) cause sounds to smear in the time domain. I mean, that’s what reverb and delay are for.

But as such, they can step on vocal monitoring’s toes a bit.

If it isn’t a specific need for the band, it’s best to leave vocals dry in monitor world. Being able to extract linguistic information from a sound is a big contributor to the perception that something is loud enough or not. If the words are hard to pick out because they’re all running together, then there’s a tendency to run things too hot in order to compensate.

The first step with vocal monitors is to get them loud enough. That’s the key goal. After that goal is met, then you can see how far you can go in terms of making things pretty. Pretty is nice, and very desirable, but it’s not the first task or the most important one.

Unsuckifying Your Monitor Mix

Communicate well, and try not to jam too much into any one mix.

Please Remember:

The opinions expressed are mine only. These opinions do not necessarily reflect anybody else’s opinions. I do not own, operate, manage, or represent any band, venue, or company that I talk about, unless explicitly noted.

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Monitors can be a beautiful thing. Handled well, they can elicit bright-eyed, enthusiastic approbations like “I’ve never heard myself so well!” and “That was the best sounding show EVER!” They can very easily be the difference between a mediocre set and a killer show, because of how much they can influence the musicians’ ability to play as a group.

I’ve said it to many people, and I’m pretty sure I’ve said it here: As an audio-human, I spend much more time worrying about monitor world than FOH (Front Of House). If something is wrong out front, I can hear it. If something is wrong in monitor world, I won’t hear it unless it’s REALLY wrong. Or spiraling out of control.

…and there’s the issue. Bad monitor mixes can do a lot of damage. They can make the show less fun for the musicians, or totally un-fun for the musicians, or even cause so much on stage wreckage that the show for the audience becomes a disaster. On top of that, the speed at which the sound on deck can go wrong can be startlingly high. If you’ve ever lost control of monitor world, or have been a musician in a situation where someone else has had monitor world “get away” from them, you know what I mean. When monitors become suckified, so too does life.

So – how does one unsuckify (or, even better, prevent suckification of) monitor world?

Foundational Issues To Prevent Suckification

Know The Inherent Limits On The Engineer’s Perception

At the really high-class gigs, musicians and production techs alike are treated to a dedicated “monitor world” or “monitor beach.” This is an independent or semi-independent audio control rig that is used to mix the show for the musicians. There are even some cases where there are multiple monitor worlds, all run by separate people. These folks are likely to have a setup where they can quickly “solo” a particular monitor mix into their own set of in-ears, or a monitor wedge which is similar to what the musicians have. Obviously, this is very helpful to them in determining what a particular performer is hearing.

Even so, the monitor engineer is rarely in exactly the same spot as any particular musician. Consequently, if the musicians are on wedges, even listening to a cue wedge doesn’t exactly replicate the total acoustic situation being experienced by the players.

Now, imagine a typical small-venue gig. There’s probably one audio human doing everything, and they’re probably listening mostly to the FOH PA. The way that FOH combines with monitor world can be remarkably different out front versus on deck. If the engineer has a capable console, they can solo up a complete monitor mix, probably through a pair of headphones. (A cue wedge is pretty unlikely to have been set up. They’re expensive and consume space.) A headphone feed is better than nothing, but listening to a wedge mix in a set of cans only tells an operator so much. Especially when working on a drummer’s mix, listening to the feed through a set of headphones has limited utility. A guy or gal might set up a nicely balanced blend, but have no real way of knowing if that mix is even truly audible at the percussionist’s seat.

If you’re not so lucky as to have a flexible console, your audio human will be limited to soloing individual inputs.

The point is that, at most small-venue shows, an audio human at FOH can’t really be expected to know what a particular mix sounds like as a total acoustic event. Remote-controlled consoles can fix this temporarily, of course, but as soon as the operator leaves the deck…all bets are off. If you’re a musician, assume that the engineer does NOT have a thoroughly objective understanding of what you’re hearing. If you’re an audio human, make the same assumption about yourself. Having made those assumptions, be gentle with yourself and others. Recognize that anything “pre set” is just a wild guess, and further, recognize that trying to take a channel from “inaudible in a mix” to “audible” is going to take some work and cooperation.

Use Language That’s As Objective As Possible

Over the course of a career, audio humans create mental mappings between subjective statements and objective measurements. For instance, when I’m working with well-established monitor mixes, I translate requests like “Could I get just a little more guitar?” into “Could I get 3 dB more guitar?” This is a necessary thing for engineers to formulate for themselves, and it’s appropriate to expect that a pro-level operator has some ability to interpret subjective requests.

At the same time, though, it can make life much easier when everybody communicates using objective language. (Heck, it makes it easier if there’s two-way communication at all.)

For instance, let’s say you’re an audio human working with a performer on a monitor mix, and they ask you for “a little more guitar.” I strongly recommend making the change that you translate “a little more” as corresponding to, and then stating your change (in objective terms) over the talkback. Saying something like, “Okay, that’s 3 dB more guitar in mix 2” creates a helpful dialogue. If that 3 dB more guitar wasn’t enough, the stating of the change opens a door for the musician to say that they need more. Also, there’s an opportunity for the musician’s perception to become calibrated to an objective scale – meaning that they get an intuitive sense for what a certain dB boost “feels” like. Another opportunity that arises is for you and the musician to become calibrated to each other’s terminology.

Beyond that, a two-way dialogue fosters trust. If you’re working on monitors and are asked for a change, making a change and then stating what you did indicates that you are trying to fulfill the musician’s wishes. This, along with the understanding that gets built as the communication continues, helps to mentally place everybody on the same team.

For musicians, as you’re asking for changes in your monitor mixes, I strongly encourage you to state things in terms of a scale that the engineer can understand. You can often determine that scale by asking questions like, “What level is my vocal set at in my mix?” If the monitor sends are calibrated in decibels, the engineer will probably respond with a decibel number. If they’re calibrated in an arbitrary scale, then the reply will probably be an arbitrary number. Either way, you will have a reference point to use when asking for things, even if that reference point is a bit “coarse.” Even if all you’ve got is to request that something go from, say, “five to three,” that’s still functionally objective if the console is labeled using an arbitrary scale.

For decibels, a useful shorthand to remember is that 3 dB should be a noticeable change in level for something that’s already audible in your mix. “Three decibels” is a 2:1 power ratio, although you might personally feel that “twice as loud” is 6 dB (4:1) or even 10 dB (10:1).

Realtime Considerations To Prevent And Undo Suckification

Too Much Loop Gain, Too Much Volume

Any instrument or device that is substantially affected by the sound from a monitor wedge, and is being fed through that same wedge, is part of that mix’s “loop gain.” Microphones, guitars, basses, acoustic drums, and anything else that involves body or airborne resonance is a factor. When their output is put through a monitor speaker, these devices combine with the monitor signal path to form an acoustical, tuned circuit. In tuned circuits, the load impedance determines whether the circuit “rings.” As the load impedance drops, the circuit is more and more likely to ring or resonate for a longer time.

If that last bit made your eyes glaze over, don’t worry. The point is that more gain (turning something up in the mix) REDUCES the impedance, or opposition, to the flow of sound in the loop. As the acoustic impedance drops, the acoustic circuit is more likely to ring. You know, feed back. *SQEEEEEALLLL* *WHOOOOOwoowooooOOOM*


The thing for everybody to remember – audio humans and musicians alike – is that a monitor mix feeding a wedge becomes progressively more unstable as gain is added. As ringing sets in, the sound quality of the mix drops off. Sounds that should start and then stop quickly begin to “smear,” and with more gain, certain frequency ranges become “peaky” as they ring. Too much gain can sometimes begin to manifest itself as an overall tone that seems harsh and tiring, because sonic energy in an irritating range builds up and sustains itself for too long. Further instability results in audible feedback that, while self-correcting, sounds bad and can be hard for an operator to zero-in on. As instability increases further, the mix finally erupts into “runaway” feedback that’s both distracting and unnerving to everyone.

The fix, then is to keep each mix’s loop gain as low as possible. This often translates into keeping things OUT of the monitors.

As an example, there’s a phenomenon I’ve encountered many times where folks start with vocals that work…and then add a ton of other things to their feed. These other sources are often far more feedback resistant than their vocal mic can be, and so they can apply enough gain to end up with a rather loud monitor mix. Unfortunately, they fall in love with the sound of that loud mix, except for the vocals which have just been drowned. As a result, they ask for the vocals to be cranked up to match. The loop gain on the vocal mic increases, which destabilizes the mix, which makes monitor world harder to manage.

As an added “bonus,” that blastingly loud monitor mix is often VERY audible to everybody else on stage, which interferes with their mixes, which can cause everybody else to want their overall mix volume to go up, which increases loop gain, which… (You get the idea.)

The implication is that, if you’re having troubles with monitors, a good thing to do is to start pulling things out of the mixes. If the last thing you did before monitor world went bad was, say, adding gain to a vocal mic, try reversing that change and then rebuilding things to match the lower level.

And not to be harsh or combative, but if you’re a musician and you require high-gain monitors to even play at all, then what you really have is an arrangement, ensemble, ability, or equipment problem that is YOURS to fix. It is not an audio-human problem or a monitor-rig problem. It’s your problem. This doesn’t mean that an engineer won’t help you fix it, it just means that it’s not their ultimate responsibility.

Also, take notice of what I said up there: High-GAIN monitors. It is entirely possible to have a high-gain monitor situation without also having a lot of volume. For example, 80 dB SPL C is hardly “rock and roll” loud, but getting that output from a person who sings at the level of a whisper (50 – 60 dB SPL C) requires 20 – 30 dB of boost. For the acoustical circuits that I’ve encountered in small venues, that is definitely a high-gain situation. Gain is the relative level increase or decrease applied to a signal. Volume is the output associated with a signal level resultant from gain. They are related to each other, but the relationship isn’t fixed in terms of any particular gain setting.

Conflicting Frequency Content

Independent of being in a high-gain monitor conundrum, you can also have your day ruined by masking. Masking is what occurs when two sources with similar frequency content become overlaid. One source will tend to dominate the other, and you lose the ability to hear both sources at once. I’ve had this happen to me on numerous occasions with pianists and guitar players. They end up wanting to play at the same time, using substantially the same notes, and the sonic characteristics of the two instruments can be surprisingly close. What you get is either too-loud guitar, too-loud piano, or an indistinguishable mash of both.

In a monitor-mix situation, it’s helpful to identify when multiple sources are all trying to occupy the same sonic space. If sources can’t be distinguished from one another until one sound just gets obliterated, then you may have a frequency-content collision in progress. These collisions can result in volume wars, which can lead to high-gain situations, which result in the issues I talked about in the previous section. (Monitor problems are vicious creatures that breed like rabbits.)

After being identified, frequency-content issues can be solved in a couple of different ways. One way is to use equalization to alter the sonic content of one source or another. For instance, a guitar and a bass might be stepping on each other. It might be decided that the bass sound is fine, but the guitar needs to change. In that case, you might end up rolling down the guitar’s bottom end, and giving the mids a push. Of course, you also have to decide where this change needs to take place. If everything was distinct before the monitor rig got involved, then some equalization change from the audio human is probably in order. If the problem largely existed before any monitor mixes were established, then the issue likely lies in tone choice or song arrangement. In that case, it’s up to the musicians.

One thing to be aware of is that many small-venue mix rigs have monitor sends derived from the same channel that feeds FOH. While this means that the engineer’s channel EQ can probably be used to help fix a frequency collision, it also means that the change will affect the FOH mix as well. If FOH and monitor world sound significantly different from each other, a channel EQ configuration that’s correct for monitor world may not be all that nice out front. Polite communication and compromise are necessary from both the musicians and the engineer in this case. (Certain technical tricks are also possible, like “multing” a problem source into a monitors-only channel.)

Lack Of Localization

Humans have two ears so that we can determine the location and direction of sounds. In music, one way for us to distinguish sources is for us to recognize those instruments as coming from different places. When localization information gets lost, then distinguishing between sources requires more separation in terms of overall volume and frequency content. If that separation isn’t possible to get, then things can become very muddled.

This relates to monitors in more than one way.

One way is a “too many things in one place that’s too loud” issue. In this instance, a monitor mix gets more and more put in it, and at a high enough volume that the monitor obscures the other sounds on deck. What the musician originally heard as multiple, individually localized sources is now a single source – the wedge. The loss of localization information may mean that frequency-content collisions become a problem, which may lead to a volume-war problem, which may lead to a loop-gain problem.

Another possible conundrum is “too much volume everywhere.” This happens when a particular source gets put through enough wedges at enough volume for it to feel as though that single source is everywhere. This can ruin localization for that particular source, which can also result in the whole cascade of problems that I’ve already alluded to.

Fixing a localization problem pretty much comes down having sounds occupy their own spatial point as much as possible. The first thing to do is to figure out if all the volume used for that particular source is actually necessary in each mix. If the volume is basically necessary, then it may be feasible to move that volume to a different (but nearby) monitor mix. For some of the players, that sound will get a little muddier and a touch quieter, but the increase in localization may offset those losses. If the volume really isn’t necessary, then things get much easier. All that’s required is to pull back the monitor feeds from that source until localization becomes established again.

It’s worth noting that “extreme” cases are possible. In those situations, it may be necessary to find a way to generate the necessary volume from a single, localized source that’s audible to everyone on the deck. A well placed sidefill can do this, and an instrument amplifier in the correct position can take this role if a regular sidefill can’t be conjured up.

Wrapping Up

This can be a lot to take in, and a lot to think about. I will freely confess to not always having each of these concepts “top of mind.” Sometimes, audio turns into a pressure situation where both musicians and techs get chased into corners. It can be very hard for a person who’s not on deck to figure out what particular issue is in effect. For folks without a lot of technical experience who play or sing, identifying a problem beyond “something’s not right” can be too much to ask.

In the heat of the moment, it’s probably best to simply remember that yes, monitors are there to be used – but not to be overused. Effective troubleshooting is often centered around taking things out of a misbehaving equation until the equation begins to behave again. So, if you want to unsuckify your monitors, try getting as much out of them as possible. You may be surprised at what actually ends up working just fine.

Mysteriously Clean

“Clean sound” has to do with more than just volume. Where that volume goes is also important.

Please Remember:

The opinions expressed are mine only. These opinions do not necessarily reflect anybody else’s opinions. I do not own, operate, manage, or represent any band, venue, or company that I talk about, unless explicitly noted.

PA030005Want to use this image for something else? Great! Click it for the link to a high-res or resolution-independent version.

So – you might be wondering what that picture of V-drum cymbals has to do with all this. I’ll gladly tell you.

Just a couple of weeks ago, the band Sake Shot was playing at my regular gig. They were the opening act, and the drummer decided that the changeover would be facilitated by the simplicity and speed of just pulling his E-kit off the deck.

During Sake Shot’s set, Brian from The Daylates walked up to FOH (Front Of House) control. After saying hello, he made a single comment that caused me to do some thinking. What he said was: “The drums sound great. It’s so clean!”

He was absolutely correct, of course. The drums were very clear, and highly separated from the other sources on stage. If the sound of the drums had been a photograph, the image would have been razor sharp. The question was, “Why?” It wasn’t just volume. The mix was somewhat quieter than some other rock bands I’ve done, but we were definitely louder than a jazz trio playing a hotel lobby (if you get my drift). No…there were other factors in play besides how much SPL (Sound Pressure Level) was involved.

I’ll start out by putting it this way: It’s not just how much volume there is. It’s also about where that volume goes.

Let me explain.

Drums, Drums, Everywhere

If you were to take a measurement microphone and walk around an acoustic drumkit, I’m reasonably sure that the overall plot of SPL levels would look something like this:


Behind the drummer, you might lose about 6 dB (or maybe not even that much), but overall, the drums just go everywhere. Sound POURS from the kit in all directions. In other words, the drumkit is NOT directional in any real way. This has a number of consequences:

1) Sound (and LOTS of it) travels forward from the kit, into the most sensitive part of the downstage vocal mics’ polar patterns. What’s wanted in those vocal mics is, of course, vocals. Anything that isn’t vocals that makes it into the mic is “noise,” which partially washes out the desired vocal signal.

2) The same sound that just hit the vocal mics continues forward to arrive at the ears of the audience.

3) That same sound also travels through the PA, courtesy of the vocal mics. Especially in a system that uses digital processing of some kind, latency is introduced. The sonic event being reproduced by the PA arrives slightly later than the acoustical event.

4) The sound traveling in directions other than straight towards the audience is – in a small venue – extremely likely to meet some sort of boundary. Some of these boundaries may have significant acoustical absorption qualities, and some of them may have almost no absorption at all. The boundaries that mostly act as reflectors (hard walls, hard ceilings, hard floors, etc) cause the sound to re-emit into the room, and that re-emitted sound can travel into the audience’s ears. These reflections also arrive later than the direct acoustical radiation from the kit. The reflections may exist in the closely packed, smooth wash of reverberation, or they might manifest as distinct “slaps” or “flutter.”

The upshot is that you have sonic events with multiple arrivals. One particular snare hit makes several journeys to the ears of the audience members, and what would otherwise be a nice, clean “crack” becomes smeared in time to some extent. Each drum transient gets sonically blurred, which means inter and intra-drum events become harder to discern from each other. (Inter-drum events are hits on different drums, whereas intra-drum events are the beginnings and ends of sounds produced by one hit on one drum.)

In short, the reflected sound of the drumkit partially garbles the direct sound of the kit. On top of that, the drum sound is now partially garbling the vocals.

This isn’t necessarily a disaster. Bands and techs deal with it all the time, and it’s possible to get perfectly acceptable sonics with an acoustic drumkit in a small venue. The point of this article isn’t to sell electronic drums to everybody. Even so, the effects of an acoustic kit’s sound careening around a room can’t be ignored.

Directivity Matters

Now then.

What was different enough about Sake Shot’s set to make Brian say that the sound was really clean?

It really wasn’t the SPL involved. When it came right down to it, the monitor rig and PA system were creating enough level to make the V-drums sound reasonably like a regular kit. The key was where that SPL was going…directivity, in other words.

Most pro-audio loudspeakers are far more directional than a drumkit. Sure, if you walk around the back of a PA speaker, you’ll still hear something. Even so, the amount of “spill” is enormously reduced. Here’s my estimate of what the average SPL coverage of an “affordable, garden-variety” pro-audio box looks like.


This is exceptionally important in the context of my regular gig, because the upstage and stage-right walls, along with a portion of the stage ceiling, are acoustically treated. Not only do the downstage monitors fire into the parts of the vocal mic patterns that are LEAST sensitive, they also fire into a boundary which is highly absorptive. Further, the drum monitors fire into the drummer’s ears, and partially into the absorptive back wall. There’s a lot less spill that can hit the reflective boundaries in the room.

What this means is that the non-direct arrivals of the E-kit’s sounds were – relative to an acoustic kit – very low in relation to the direct arrivals from the FOH PA. Further, there was very little “wash” in the vocal mics. All this added up to a sound that was very clean and defined, because each transient from the drums had a sharply defined beginning and end. This makes it much easier for a listener to figure out where drum sounds stop, and where other things (like vocal consonants) begin. Further, the vocal mics were generally delivering a rather higher signal-to-noise ratio than they otherwise might have been, which cleaned up the vocals AND the sound of the drums.

All the different sounds from the show were doing a lot less “running into each other.”

As such, the mysteriously clean sound of the show wasn’t so mysterious after all.

Speed Fishing

“Festival Style” reinforcement means you have to go fast and trust the musicians.

Please Remember:

The opinions expressed are mine only. These opinions do not necessarily reflect anybody else’s opinions. I do not own, operate, manage, or represent any band, venue, or company that I talk about, unless explicitly noted.

Want to use this image for something else? Great! Click it for the link to a high-res or resolution-independent version.

Last Sunday was the final day of the final iteration of a local music festival called “The Acoustic All-Stars.” It’s a celebration of music made with traditional or neo-traditional instruments – acoustic-electric guitars, fiddles, drums, mandolins, and all that sort of thing. My perception is that the musicians involved have a lot of anticipation wrapped up in playing the festival, because it’s a great opportunity to hear friends, play for friends, and make friends.

Of course, this anticipation can create some pressure. Each act’s set has a lot riding on it, but there isn’t time to take great care with any one setup. The longer it takes to dial up the band, the less time they have to play…and there are no “do overs.” There’s one shot, and it has to be the right shot for both the listeners and the players.

The prime illustrator for all this on Sunday was Jim Fish. Jim wanted to use his slot to the fullest, and so assembled a special team of musicians to accompany his songs. The show was clearly a big deal for him, and he wanted to do it justice. Trying to, in turn, do justice to his desires required that a number of things take place. It turns out that what had to happen for Jim can (I think) be generalized into guidelines for other festival-style situations.

Pre-Identify The Trouble Spots, Then Make The Compromises

The previous night, Jim had handed me a stage plot. The plot showed six musicians, all singing, wielding a variety of acoustic or acoustic-electric instruments. A lineup like that can easily have its show wrecked by feedback problems, because of the number of open mics and highly-resonant instruments on the deck. Further, the mics and instruments are often run at (relatively) high-gain. The PA and monitor rig need to help with getting some more SPL (Sound Pressure Level) for both the players and the audience, because acoustic music isn’t nearly as loud as a rock band…and we’re in a bar.

Also, there would be a banjo on stage right. Getting a banjo to “concert level” can be a tough test for an audio human, depending on the situation.

Now, there’s no way you’re going to get “rock” volume out of a show like this – and frankly, you don’t want to get that kind of volume out of it. Acoustic music isn’t about that. Even so, the priorities were clear:

I needed a setup that was based on being able to run with a total system gain that was high, and that could do so with as little trouble as possible. As such, I ended up deploying my “rock show” mics on the deck, because they’re good for getting the rig barking when in a pinch. The thing with the “rock” mics is that they aren’t really sweet-sounding transducers, which is unfortunate in an acoustic-country situation. A guy would love to have the smoothest possible sound for it all, but pulling that off in a potentially high-gain environment takes time.

And I would not have that time. Sweetness would have to take a back seat to survival.

Be Ready To Abandon Bits Of The Plan

On the day of the show, the lineup ended up not including two people: The bassist and the mandolin player. It was easy to embrace this, because it meant lower “loop gain” for the show.

I also found out that the fiddle player didn’t want to use her acoustic-electric fiddle. She wanted to hang one particular mic over her instrument, and then sing into that as well. We had gone with a similar setup at a previous show, and it had definitely worked. In this case, though, I was concerned about how it would all shake out. In the potentially high-gain environment we were facing, pointing this mic’s not-as-tight polar pattern partially into the monitor wash held the possibility for creating a touchy situation.

Now, there are times to discuss the options, and times to just go for it. This was a time to go for it. I was working with a seasoned player who knew what she wanted and why. Also, I would lose one more vocal mic, which would lower the total loop-gain in the system and maybe help us to get away with a different setup. I knew basically what I was getting into with the mic we chose for the task.

And, let’s be honest, there were only minutes to go before the band’s set-time. Discussing the pros and cons of a sound-reinforcement approach is something you do when you have hours or days of buffer. When a performer wants a simple change in order to feel more comfortable, then you should try to make that change.

That isn’t to say that I didn’t have a bit of a backup plan in mind in case things went sideways. When you’ve got to make things happen in a hurry, you need to be ready to declare a failing option as being unworkable and then execute your alternate. In essence, festival-style audio requires an initial plan, some kind of backup plan, the willingness to partially or completely drop the original plan, and an ability to formulate a backup plan to the new plan.

The fiddle player’s approach ended up working quite nicely, by the way.

Build Monitor World With FOH Open

If there was anything that helped us pull-off Jim’s set, it was this. In a detail-oriented situation, it can be good to start with your FOH (Front Of House) channels/ sends/ etc. muted (or pulled back) while you build mixes for the deck. After the monitors are sorted out, then you can carefully fill in just what you need to with FOH. There are times, though, that such an approach is too costly in terms of the minutes that go by while you execute. This was one such situation.

In this kind of environment, you have to start by thinking not in terms of volume, but in terms of proportions. That is, you have to begin with proportions as an abstract sort of thing, and then arrive at a workable volume with all those proportions fully in effect. This works in an acoustic music situation because the PA being heavily involved is unlikely to tear anyone’s head off. As such, you can use the PA as a tool to tell you when the monitor mixes are basically balanced amongst the instruments.

It works like this:

You get all your instrument channels set up so that they have equal send levels in all the monitors, plus a bit of a boost in the wedge that corresponds to that instrument’s player. You also set their FOH channel faders to equal levels – probably around “unity” gain. At this point, the preamp gains should be as far down as possible. (I’m spoiled. I can put my instruments on channels with a two-stage preamp that lets me have a single-knob global volume adjustment from silence to “preamp gain +10 dB.” It’s pretty sweet.)

Now, you start with the instrument that’s likely to have the lowest gain before feedback. You begin the adventure there because everything else is going to have to be built around the maximum appropriate level for that source. If you start with something that can get louder, then you may end up discovering that you can’t get a matching level from the more finicky channel without things starting to ring. Rather than being forced to go back and drop everything else, it’s just better to begin with the instrument that will be your “limiting factor.”

You roll that first channel’s gain up until you’ve got a healthy overall volume for the instrument without feedback. Remember, both FOH and monitor world should both be up. If you feel like your initial guess on FOH volume is blowing past the monitors too much (or getting swamped in the wash), make the adjustment now. Set the rest of the instruments’ FOH faders to that new level, if you’ve made a change.

Now, move on to the subsequent instruments. In your mind, remember what the overall volume in the room was for the first instrument. Roll the instruments’ gains up until you get to about that level on each one. Keep in mind that what I’m talking about here is the SPL, not the travel on the gain knob. One instrument might be halfway through the knob sweep, and one might be a lot lower than that. You’re trying to match acoustical volume, not preamp gain.

When you’ve gone through all the instruments this way, you should be pretty close to having a balanced instrument mix in both the house and on deck. Presetting your monitor and FOH sends, and using FOH as an immediate test of when you’re getting the correct proportionality is what lets you do this.

And it lets you do it in a big hurry.

Yes, there might be some adjustments necessary, but this approach can get you very close without having to scratch-build everything. Obviously, you need to have a handle on where the sends for the vocals have to sit, and your channels need to be ready to sound decent through both FOH and monitor-world without a lot of fuss…but that’s homework you should have done beforehand.

Trust The Musicians

This is probably the nail that holds the whole thing together. Festival-style (especially in an acoustic context) does not work if you aren’t willing to let the players do their job, and my “get FOH and monitor world right at the same time” trick does NOT work if you can’t trust the musicians to know their own music. I generally discourage audio humans from trying to reinvent a band’s sound anyway, but in this kind of situation it’s even more of something to avoid. Experienced acoustic music players know what their songs and instruments are supposed to sound like. When you have only a couple of minutes to “throw ‘n go,” you have to be able to put your faith in the music being a thing that happens on stage. The most important work of live-sound does NOT occur behind a console. It happens on deck, and your job is to translate the deck to the audience in the best way possible.

In festival-style acoustic music, you simply can’t “fix” everything. There isn’t time.

And you don’t need to fix it, anyway.

Point a decent mic at whatever needs micing, put a working, active DI on the stuff that plugs in, and then get out of the musicians’ way.

They’ll be happier, you’ll be happier, you’ll be much more likely to stay on schedule…it’s just better to trust the musicians as much as you possibly can.