Tag Archives: EQ

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.

Projector Hangers

Just throwing a bunch of sound into a room is NOT pro-audio.

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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Since pre-school, I have known that calling people names is not a very nice thing. It’s not something that Winnie The Pooh would do, except by mistake, and if he did make that mistake, the end of the episode would have all the residents of the Hundred-Acre Wood coming together to learn a very special lesson. Eeyore would say something unintentionally funny. Christopher Robin would say “Silly old bear.”

But I sometimes do invent derogatory names for people and organizations. I especially invent those names when people or organizations manage to get things wrong in a very complete and glaring way – particularly when it comes to live audio.

(In the name of fairness, let me present that audio-humans are sometimes called “Squeaks.” It’s a reference to some of the unpleasant sounds we’ve been known to make, which includes feedback. I believe I have earned the label on more than one occasion.)


Events over the last couple of weeks have led me to concoct the epithet, “Projector Hanger.” A Projector Hanger may also be called a “D!@# Projector Hanger,” or a “G!@ D!@# Projector Hanger,” or even a “F!@#$%^ PROJECTOR HANGER,” depending upon just how much of a metaphorical mess they’ve left for folks like me to clean up.


A Projector Hanger is an A/V integrator who has no business installing a public-address audio system (because they have no clue about what makes such a system actually work well), yet installs such a system anyway. They attach this fundamentally screwed-up monstrosity to the bit of the install they actually do understand: A reasonably bright projector, maybe with HD capability, which is pointed at a screen and supplied with various inputs at some convenient location. This final bit of behavior actually provides a gateway to identifying – and hopefully avoiding – Projector Hangers.


Projector Hangers are adept at directly firing some sort of light emitter at a reflective target, such as a proper screen or brightly-painted wall. They seem to assume that this is the way to go for everything, and so they have an alarming tendency to fire sound emitters (loudspeakers) directly at reflective targets. Reflective targets like…hardwood floors. This creates an acoustical crapstorm of multiple, secondary arrivals, ensuring that everybody in the room is sitting as deeply in a reverberant field as may be practicable. Intelligibility drops like a rock as transients from words spoken into a microphone smear, bounce, ricochet, and rattle to the maximum extent possible. Gain before feedback throws up its hands and takes a sick day as overhead loudspeakers fire into the sides of microphones, and also as those microphones pick up even more re-entrant noise from the vortex of acoustical reflections.

(A primary indicator of a Projector Hanger is that the audio side of the system LOOKS nice, being unobtrusive and able to blend in with the decor, but the actual audio from the audio side SOUNDS awful.)

More Is Better, Right?

The Projector Hanger is a lover of large images. Wide throw. Multiple screens. Make sure everyone can see it! They apply this same mentality to audio, seemingly thinking that the key to everyone hearing well is for everyone to just hear something. Anything!

To accomplish this, the Projector Hanger installs a lot of speakers, with the intent that sound should be sprayed everywhere. So, even before the sound from all those loudspeakers smacks into the floor, a nightmare of multiple arrivals and destructive interference has been summoned. Also, the Projector Hanger can be counted on to compound this problem by deploying loudspeakers in spaced pairs. (The ability to reproduce stereo sound from a playback device is paramount, even if the critical application for the system is to reinforce the signal from a single microphone.) These spaced pairs further aggravate the multiple arrival and interference problems, and also feed the gaping maw of the acoustical issues: Why just hit the floor with a bunch of sound when you can also hit the walls!

Math Is Hard

Another indicator that a Projector Hanger has been on the loose is when equal numbers no longer correspond. For example:

The Projector Hanger, wishing to be helpful, installs an easy-access XLR jack for a microphone line. The jack is labeled “Mic 1,” and the label even looks like it was silkscreened directly onto the jackplate. It all looks so PRETTY.

They then permanently wire the output of that jack to a set of terminal blocks on a super-classy input mixer and amplifier. The control knobs on the device were clearly labeled at the factory, so that a person could easily find the gain controls for various channels. It would make sense, then, that the jackplate labeled “Mic 1” would be wired to “Input 1” on the mixer-amplifier unit. Of course, that’s not what happens. In an astounding bit of mental gymnastics, perhaps influenced by the literary horrorscapes of HP Lovecraft, the Projector Hanger decides that “1” is actually equal to “2.” That’s where the jack is wired. Input 1 is actually used for the installed wireless system – but no labeling is put in place to clear this up.

One day, an audio-human ties into the system through “Mic 1.” All the knobs on the mixer-amplifier are down, meaning that no signal passes through the system. The audio-human rolls the volume up on “Input 1,” but no noise is heard. The audio-human naturally thinks that there’s a cabling problem, and proceeds to waste a huge chunk of time looking for the bad connection. Eventually, the sound craftsperson gives up and deploys their fallback option – only to later discover that the whole mess was caused by a moronic, undocumented connection scheme.

I can see the argument in my head right now.

Projector Hanger: “Why didn’t you try more knobs?”

Sound Person: “Why can’t YOU COUNT?”

The Backup Is Better Than The Primary

The interface (or, perhaps more appropriately, catastrophic collision) between a pro-audio tech and a Projector Hanger is highly instructive in other ways. I mentioned above that a series of problems might force an audio human to take an alternate route. This alternate route might have been, say, patching into a single, inexpensive, powered loudspeaker sitting on a tripod stand.

Now then.

Before this particular debacle, the sound person had been trying valiantly to spray-paint the acoustical turd that the Projector Hanger had created. To this end, a very large number of parametric EQ filters had been used. By “large number,” what I mean is, “all that were available.” The EQ transfer function applied to try to make the system usable was (quite frankly) insane, and was implemented across two processors. One was patched across the total output of the audio human’s mixer, and the other was inserted on a wireless headset.

When the fallback solution was implemented, the tech bypassed all the EQ. All that crazy finagling could not be counted on to be helpful in the situation, so it was better to start from scratch. This was unfortunate, but the operator was poised and ready to fight any problems in realtime. Some faders for wired mics were pushed up on the console.

The sound was, actually, very good. With nothing beyond basic channel EQ, the single, ugly, cheap loudspeaker on a stick was handily beating the CRAP out of the multi-unit, nice-looking, expensive install. The headset mic also had plenty of usable gain, although the audio human did use a few inserted filters to clean up a bit of mud and harshness. When an emergency-implemented “I don’t know what else to do, so let’s just get through it” solution works better than the thing that was all planned out…

…you might just have had a run in with a Projector Hanger.

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.

How Powerful An Amp Should I Buy?

For safety, match the continuous ratings. For performance, match the peak ratings.

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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For people who buy passive speakers (loudspeakers driven by amplifiers in separate enclosures), the question of how much amp to purchase is somewhat sticky. Ask it, and you’ll get all manner of advice. Some of it good, some of it bad, and some of it downright ludicrous. You’re very likely to hear a bunch of hoo-ha about how using too small of an amp is dangerous (it isn’t), because clipping kills drivers (it doesn’t). Someone will eventually say that huge amps give you more headroom (sorry, but no). All kinds of “multipliers” will be bandied about.

You may become more confused than when you started.

In my opinion, the basic answer is pretty simple, although the explanation will take a bit of time:

First, note that even though physicists will tell you that there’s no such thing as “RMS power,” there IS such a thing as the average or continuous power derived from a certain RMS voltage input. That’s what “RMS power” on a spec sheet means.

For a reasonable balance of safety and performance, match the amp’s continuous rating with the loudspeaker’s continuous rating.

(If you cannot find a loudspeaker’s continuous rating, clearly stated, on a spec sheet, take the smallest rating you can find and divide by two. If you cannot easily find an amp’s continuous rating on a spec sheet, just choose a different amplifier.)

For slightly more performance at the cost of some safety, choose an amplifier with a continuous rating that is half the peak power handling of the loudspeaker.

Depending upon how a loudspeaker is rated, the “safety” and “performance” criteria may actually end up giving you the same answer. This is perfectly acceptable.

Now then. Here’s how I justify my advice.

Peak / √2

The first step here is to understand a bit more about some basic bath and science regarding amplifiers.

A power amplifier is really a voltage amplifier that can deliver enough current to drive a loudspeaker motor. A power amplifier has an upper limit to how much voltage it can develop, as you might expect. That maximum voltage, combined with the connected load and the amplifier’s ability to supply current, determines the amplifier’s peak power.

In normative cases, an amplifier’s peak output is an “instantaneous” event. If the amplifier is contributing no noticeable distortion to the signal, then the signal “swing” is reaching the amplifier’s maximum voltage for a very small amount of time. (Ideally, an infinitely small duration.) Again, if we assume normal operation, an amplifier spends the overwhelming majority of its life producing less than maximum output.

An amplifier’s continuous power, on the other hand, is an average over a significant amount of time. This is why engineers say things like “power is the area under the curve.” An undistorted peak with nothing before or after it has virtually no area under the curve, whereas a signal that never gets anywhere near peak output (but lasts for several seconds) can have very significant area under the curve.

For audio voltages, we use RMS averaging. One important reason for this is because audio voltages corresponding to sound-pressure events have positive and negative swing. For, say, one cycle of a sine wave, the arithmetic mean would be zero – the wave has equal positive and negative value. RMS averaging, on the other hand, squares each input value. As such, positive values remain positive, and negative values become positive (-2 squared, for instance, is 4).

In the case of an undistorted sine wave, the RMS voltage is the peak voltage divided by √2, or about 1.414.

Here’s a graph to make this all easier to visualize. This is a plot of a very small, hypothetical power amplifier passing an undistorted sine wave. The maximum output voltage is ± 2 volts. That means that the RMS voltage is 2/√2, or 1.414.


Here’s where the rubber begins to meet the road. Let’s assume that this amplifier is mated to a loudspeaker with an impedance of 8 Ohms.

Power = Voltage Squared / Resistance

Peak Power = Peak Voltage Squared / Resistance

Continuous Power = RMS Voltage Squared / Resistance

Peak Power = 2^2 / 8 = 0.5 Watts

Continuous Power = (2 / √2)^2 / 8 = 0.25 watts

For a sine wave, the continuous power is half the peak power, or 3 dB down. This is the main justification for the above statement: “For slightly more performance at the cost of some safety, choose an amplifier with a continuous rating that is half the peak power handling of the loudspeaker.” Assuming that the amplifier was rated using sine-wave input (a reasonable assumption at the time of this writing), the peak output of the amplifier will be twice the continuous rating, and therefore match up with the peak power handling of the loudspeaker. By the same token, the “safety” recommendation means that the peak amp output will be either at or far below the peak rating of the loudspeaker – especially since many loudspeakers are claimed to handle peaks that are four times greater than the recommended continuous input.

An amplifier with peak output capabilities that exceeds the peak handling capabilities of a loudspeaker is a liability, not an asset. In live-sound, all kinds of mishaps can occur which will drive an amp all the way to its maximum output. If that maximum output is too high, you might just have an expensive repair on your hands. If the maximum amplifier output plays nicely with the loudspeaker’s capabilities, however, accidents are much less worrisome.

So, there’s the explanation in terms of peak power. What about some other angles?

A More Holistic Picture

Musical signals running through a PA are usually not pure sine waves. They can be decomposed into pure tones, certainly, but the total signal behavior is not “RMS voltage = peak / √2.” You might have an overall continuous power level that’s 10 dB, 12 dB, 15 dB, or even farther down from the peaks. Why could you still run into problems?

The short answer is that not all drivers are created equally, and EQ can make them even more unequal. Further, EQ can cause you to be rather more unkind than you might realize.

For a bit more detail, let’s make up a compromise example using pink noise that has a crest factor of slightly more than 13 decibels. If we run the signal full-range, we get statistics that look like this:


Let’s say that we have a QSC GX5 plugged into an 8 Ohm loudspeaker. A GX5 is rated for 500 watts continuous into that load, so a reasonable guess at peak output is 1000 watts. To find -13 dB in terms of power:

10 log (x / 1000) = -13 dB

log (x / 1000) = -1.3 dB

10^-1.3 = 0.0501 = x / 1000

x = 50 watts

(Of course, -13 dB can also be found by dividing -10 dB, or 0.1 X power, by two.)

That power hits a passive crossover, which splits the full range signal into appropriate passbands for the various drivers. In an affordable, two-way box, the crossover might be something like 12 dB / octave at 2000 Hz. If I filter the noise accordingly, I get this for what the LF driver “sees”:


Compared to the original peak, the LF driver is seeing about -14.5 dB continuous, or a bit more than 35 watts. Some instantaneous levels of about 800 watts come through, but the driver can probably soak those up if most of the energy is above, say, 40 Hz.

For the HF driver:


Again, we have to compare things to the original peak of -0.89 dB, so the continuous measurement is actually 17.8 dB down from there. Also, an additional complication exists. The HF driver is probably padded down at the crossover, because a compression driver mated to a horn can have a sensitivity of 104+ dB @ 1 watt @ 1 meter, whereas the cone driver might be only 96 dB or so. In the case of an 8 dB pad, the total continuous power being experienced by the HF portion of the box could reasonably be said to be -25.8 dB from the peak power. That’s something like 2.5 watts, with peaks at 37 watts or so.

No problem, right?

But what if you bought a really powerful amp – like one that could deliver peaks of 2000 watts?

Your HF driver would still be okay, but your LF driver might not be. Sure, 70 watts continuous wouldn’t burn up the voice coil, but what would 1600 watt peaks do? Especially if the information is “down deep,” that poor cone is likely to get ripped apart. If somebody does something like dropping a mic…well…

And what if someone applies the dreaded “smiley face” EQ, and then drives the amp right up to the clip lights?

At first, things still look OK. The continuous signal is still 13 dB down from the peaks.


The LF driver is getting something like this:


For the reasonably-sized amp, the LF peaks are at 0.7 dB below clipping, or 850 watts. That’s probably a little too much for the driver, but it might not die immediately – unless a huge impulse under 40 Hz comes through. With the oversized amplifier, you now have 1700 watt peaks, which are beating up your LF cone just that much faster.

In the world of the HF driver:


Using the appropriate amp, the HF driver isn’t getting cooked at all. In fact, the abundance of LF content actually pushes the continuous and peak power down slightly. Even the big amp isn’t an issue.

Of course, someone could decide to only crank the highs, because they want “that crispness, you know?” (This would also correspond to program material that’s heavily biased towards HF information.)


Now things get a little scary. Scale the measurement right up to clipping (0 dB, because this reading was taken “in isolation”), and the peaks are padded down to only -8 dB. That’s almost 160 watts, which is beyond the peak tolerance of the driver. The 13 watts of continuous input isn’t hurting anything, but the poor little HF unit is taking plenty of abuse.

Connect the “more headroom, dude!” amplifier, and it gets much worse. One 320 watt peak will surely be enough to end the life of the unit, and if (by some miracle) the peaks are limited but the continuous power isn’t…well, the driver might withstand 26 watts continuous, but just two more dB and you get 41 watts. The poor baby is probably roasting, if it’s an affordable unit.


I’m sorry if all that caused your eyes to glaze over. Here’s how it shakes out:

An amp which has a continuous rating that matches the loudspeaker’s continuous rating does a lot to protect you from abuse, accidents, and stupidity. Using an amplifier that has a peak rating equal to the speaker’s peak rating lets you get a bit more level (3 dB) while still shielding you from a lot of problems. You can still get yourself into trouble, but it takes some effort.

Running an amplifier which goes a long way past the peak rating of a speaker enclosure is just asking for something to get wrecked. Yes, you can make it all work if you’re careful and use well-set processing to keep things sane – but that’s beyond the scope of this article.

If a conservatively powered PA doesn’t get loud enough for you, you need more PA. That is, you need more boxes powered at the same per-box level, or boxes that are naturally louder, or boxes that will take more power.

It’s Not Actually About The Best Sound

What we really want is the best possible show at the lowest practical gain.

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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As it happens, there’s a bit of a trilogy forming around my last article – the one about gain vs. stability. In discussions like this, the opening statement tends to be abstract. The “abstractness” is nice in a way, because it doesn’t restrict the application too much. If the concept is purified sufficiently, it should be usable in any applicable context.

At the same time, it’s nice to be able to make the abstract idea more practical. That is, the next step after stating the concept is to talk about ways in which it applies.

In live audio, gain is both a blessing and a curse. We often need gain to get mic-level signals up to line-level. We sometimes need gain to correct for “ensemble imbalances” that the band hasn’t yet fixed. We sometimes need gain to make a quiet act audible against a noisy background. Of course, the more gain we add, the more we destabilize the PA system, and the louder the show gets. The day-to-day challenge is to find the overall gain which lets us get the job done while maintaining acceptable system stability and sound pressure.

If this is the overall task, then there’s a precept which I think can be derived from it. It might only be derivable indirectly, depending on your point of view. Nevertheless:

Live sound is NOT actually about getting the best sound, insofar as “the best sound” is divorced from other considerations. Rather, the goal of live sound is to get the best possible holistic SHOW, at the lowest practical gain.

Fixing Everything Is A Bad Idea

The issue with a phrase like “the best sound” is that it morphs into different meanings for different people. For instance, at this stage in my career, I have basically taken the label saying “The Best Sound” and stuck it firmly on the metaphorical box containing the sound that gets the best show. For that reason alone, the semantics can be a little difficult. That’s why I made the distinction above – the distinction that “the best sound” or “the coolest sound” or “the best sound quality” is sometimes thought of without regard to the show as a whole.

This kind of compartmentalized thinking can be found both in concert audio veterans and greenhorns. My gut feeling is that the veterans who still section off their thinking are the ones who never had their notions challenged when they were new enough.

…and I think it’s quite common among new audio humans to think that the best sound creates the best show. That is, if we get an awesome drum sound, and a killer guitar tone, and a thundering bass timbre, and a “studio ready” vocal reproduction, we will then have a great show.

The problem with this line of thinking is that it tends to create situations where a tech is trying to “fix” almost everything about the band. The audio rig is used as a tool to change the sound of the group into a processed and massaged version of themselves – a larger than life interpretation. The problem with turning a band into a “bigger than real” version of itself is that doing so can easily require the FOH PA to outrun the acoustical output of the band AND monitor world by 10 dB or more. Especially in a small-venue context, this can mean lots and lots of gain, coupled with a great deal of SPL. The PA system may be perched on the edge of feedback for the duration of the show, and it may even tip over into uncontrolled ringing on occasion. Further, the show can easily be so loud that the audience is chased off.

To be blunt, your “super secret” snare-drum mojo is worthless if nobody wants to be in the same room with it. (If you follow me.)

Removed from other factors, the PA does sound great…but with the other factors being considered, that “great” sound is creating a terrible show.


The correction for trying to fix everything is to only reinforce what actually needs help. This approach obeys the “lowest possible gain” rule. PA system gain is applied only to the sources that are being acoustically swamped, and only in enough quantity that those sources stop being swamped.

In a sense, you might say that there’s a certain amount of total gain (and total resultant volume) that you can have that is within an acceptable “window.” When you’ve used up your allotted amount of gain and volume, you need to stop there.

At first, the selectivity of what gets gain applied is not very narrow. For newer operators and/ or simplified PA systems, the choice tends to be “reproduce most of the source or none of it.” You might have, say, one guitar that’s in the PA, plus a vocal that’s cranked up, and some kick drum, and that’s all. Since the broadband content of the source is getting reproduced by the PA, adding any particular source into the equation chews up your total allowable gain in a fairly big hurry. This limits the correction (if actually necessary) that the PA system can apply to the total acoustical solution.

The above, by the way, is a big reason why it’s so very important for bands to actually sound like a band without any help from the PA system. That does NOT mean “so loud that the PA is unnecessary,” but rather that everything is audible in the proper proportions.


As an operator learns more and gains more flexible equipment, they can be more selective about what gets a piece of the gain allotment. For instance, let’s consider a situation where one guitar sound is not complementing another. The overall volumes are basically correct, but the guitar tones mask each other…or are masked by something else on stage. An experienced and well-equipped audio human might throw away everything in one guitar’s sound, except for a relatively narrow area that is “out of the way” of the other guitar. The audio human then introduces just enough of that band-limited sound into the PA to change the acoustical “solution” for the appropriate guitar. The stage volume of that guitar rig is still producing the lion’s share of the SPL in the room. The PA is just using that SPL as a foundation for a limited correction, instead of trying to run right past the total onstage SPL. The operator is using granular control to get a better show (where the guitars each have their own space) while adding as little gain and SPL to the experience as possible.

If soloed up, the guitar sound in the PA is terrible, but the use of minimal gain creates a total acoustical solution that is pleasing.

Of course, the holistic experience still needs to be considered. It’s entirely possible to be in a situation that’s so loud that an “on all the time” addition of even band-limited reinforcement is too much. It might be that the band-limited channel should only be added into the PA during a solo. This keeps the total gain of the show as low as is practicable, again, because of granularity. The positive gain is restricted in the frequency domain AND the time domain – as little as possible is added to the signal, and that addition is made as rarely as possible.

An interesting, and perhaps ironic consequence of granularity is that you can put more sources into the PA and apply more correction without breaking your gain/ volume budget. Selective reproduction of narrow frequency ranges can mean that many more channels end up in the PA. The highly selective reproduction lets you tweak the sound of a source without having to mask all of it. You might not be able to turn a given source into the best sound of that type, but granular control just might let you get the best sound practical for that source at that show. (Again, this is where the semantics can get a little weird.)

Especially for the small-venue audio human, the academic version of “the best sound” might not mean the best show. This also goes for the performers. As much as “holy grail” instrument tones can be appreciated, they often involve so much volume that they wreck the holistic experience. Especially when getting a certain sound requires driving a system hard – or “driving” an audience hard – the best show is probably not being delivered. The amount of signal being thrown around needs to be reduced.

Because we want the best possible show at the lowest practical gain.

Why Broad EQ Can’t Save You

You can’t do microsurgery with an axe.

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 don’t have anything against “British EQ” as a basic concept. As I’ve come to interpret it, “British EQ” is a marketing term that means “our filters are wide.” EQ filters with gentle, wide slopes tend to sound nice and are pretty easy to use, so they make sense as a design decision in consoles that can’t give you every bell and whistle.

When I’m trying to give a channel or group a push in a specific area, I do indeed prefer to use a filter that’s a bit wider. Especially if I have to really “get on the gas,” I need the EQ to NOT impart a strange or ugly resonance to the sound. Even so, I think my overall preference is still for a more focused filter than what other folks might choose. For instance, when adding 6 dB at about 1kHz to an electric guitar (something I do quite often), the default behavior of my favorite EQ plugin is a two-octave wide filter:


What I generally prefer is a 1.5-octave filter, though.


I still mostly avoid a weird, “peaky” sound, but I get a little bit less (1 dB) of that extra traffic at 2-3 kHz, which might be just enough to keep me from stomping on the intelligibility of my vocal channels.

Especially in the rough-and-tumble world of live audio, EQ selectivity is a big deal. When everything is bleeding into everything else, you want to be able to grab and move only the frequency range that corresponds to what’s actually “signal” in a channel. Getting what you want…and also glomming onto a bunch of extra material isn’t all that helpful. In the context of, say, a vocal mic, only the actual vocal part is signal. Everything else is noise, even if it’s all music in the wider sense. IF I want to work on something in a vocal channel, I don’t also want to be working on the bass, drums, guitar, and keyboard noises that are also arriving at the mic. Selective EQ helps with that.

What Your Channel EQ Is Doing To You

Selective EQ isn’t always a choice that you get, though. If a console manufacturer has a limited “budget” to decide what to give you on a channel-per-channel basis, they’ll probably choose a filter that’s fairly wide. For instance, here’s a 6 dB boost at 1 kHz on a channel from an inexpensive analog console (a Behringer SL2442):


The filter looks to be between 2.5 and 3 octaves wide. This is perfectly fine for basic tone shaping, but it’s not always great for solving problems. It would be nice to get control over the bandwidth of the filter, but that option chews up both what can be spent on internal components, and it also hogs control-surface real estate. For those reasons, and also because of “ease of use” considerations, fully parametric EQ isn’t something that’s commonly found on small-venue, analog consoles. As such, their channel EQs are often metaphorical axes – or kitchen knives, if you’re lucky – when what you may need is a scalpel.

If you need to do something drastic in terms of gain, a big, fat EQ filter can start acting like a volume control across the entire channel. This is especially true when you need to work on two or more areas, and multiple filters overlap. You can kill your problem, but you’ll also kill everything else.

It’s like getting rid of a venomous spider by having the Air Force bomb your house.

I should probably stop with the metaphors…

Fighting Feedback

Of course, we don’t usually manage feedback issues with a console’s channel EQ. We tend to use graphic EQs that have been inserted or “inlined” on console outputs. (I do things VERY differently, but that’s not the point of this article.)

Why, though? Why use a graphic EQ, or a highly flexible parametric EQ for battling feedback?

Well, again, the issue is selectivity.

See, if what you’re trying to do is to maximize the amount of gain that can be applied to a system, any gain reduction works against that goal.

(Logical, right?)

Unfortunately, most feedback management is done by applying negative gain across some frequency range. The trick, then, is to apply that negative gain across as narrow a band as is practicable. The more selective a filter is, the more insane things you can do with its gain without having a large effect on the average level of the rest of the signal.

For example, here’s a (hypothetical) feedback management filter that’s 0.5 octaves wide and set for a gain of -9 dB.


It’s 1 dB down at about 600 Hz and 1700 Hz. That’s not too bad, but take a look at this quarter-octave notch filter:


Its actual gain is negative infinity, although the analyzer “only” has enough resolution to show a loss of 30 dB. (That’s still a very deep cut.) Even with a cut that displays as more than three times as deep as the first filter, the -1 dB points are 850 Hz and 1200 Hz. The filter’s high selectivity makes it capable of obliterating a problem area while leaving almost everything else untouched.

To conclude, I want to reiterate: Wide EQ isn’t bad. It’s an important tool to have in the box. At the same time, I would caution craftspersons that are new to this business that a label like “British EQ” or “musical EQ” does not necessarily mean “good for everything.” In most cases, what that label likely means is that an equalizer is inoffensive by way of having a gentle slope.

And that’s fine.

But broad EQ can’t save you. Not from the really tough problems, anyway.

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.

Echoes Of Feedback

By accident, I seem to have discovered an effective, alternate method for “ringing out” PA systems and monitor rigs.

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.

Sometimes, the best way to find something is to be looking for something else entirely.

A couple of weeks ago, I got it into my head to do a bit of testing. I wanted to see how much delay time I could introduce into a monitor feed before I noticed that something was amiss. To that end, I took a mic and monitor that were already set up, routed the mic through the speaker, and inserted a delay (with no internal feedback) on the signal path. I walked between FOH (Front Of House) and the stage, each time adding another millisecond of delay and then talking into the mic.

For several go-arounds, everything was pretty nondescript. I finally got to a delay time that was just noticeable, and then I thought, “What the heck. I should put in something crazy to see how it sounds.” I set the delay time to something like a full second, and then barked a few words into the mic.

That’s when it happened.

First, silence. Then, loud and clear, the delayed version of what I had said.

…and then, the delayed version of the delayed version of what I had just said, but rather more quietly.

“Whoops,” I thought, “I must have accidentally set the delay’s feedback to something audible.” I began walking back to FOH, only to suddenly realize that I hadn’t messed up the delay’s settings at all. I had simply failed to take into account the entire (and I do mean the ENTIRE) signal path I was working on.

Hold that thought.

Back In The Day

There was a time when delay effects weren’t the full-featured devices we’re used to. Whether the unit was using a bit of tape or some digital implementation, you didn’t always get a processor with a knob labeled “feedback,” or “regen,” or “echoes,” or whatever. There was a chance that your delay processor did one thing: It made audio late. Anything else was up to you.

Because of this, certain consoles of the day had a feature on their aux returns that allowed for the signal passing through the return to be “multed” (split), and then sent back through the aux send to the processor it came from. (On SSL consoles, this feature was called “spin.”) You used this to get the multiple echoes we usually associate with delay as an effect for vocals or guitar.

At some point, processor manufacturers decided that including this feature inside the actual box they were selling was a good idea, and we got the “feedback” knob. There’s nothing exotic about the control. It just routes some of the output back to the input. So, if you have a delay set for some number of milliseconds, and send a copy of the output back to the input end (at a reduced level), then you get a repeat every time your chosen number of milliseconds ticks by. Each repeat drops in level by the gain reduction applied at the feedback control…and eventually, the echo signal can’t be readily heard anymore.

But anyway, the key point here is that whether or not it’s handled “internally,” repeating echoes from a delay line are usually caused by some amount of the processor’s output returning to the front end to be processed again. (I say “usually” because it’s entirely possible to conceive of a digital unit that operates by taking an input sample, delaying the sample, playing the sample back at some volume, and then repeats the process for the sample a certain number of times before stopping the process. In this case, the device doesn’t need to listen to its own output to get an echo.)

I digress. Sorry.

If the output were to be routed back to the input at “unity gain,” (with no reduction or increase in level relative to the original output signal) what would happen? That’s right – you’d get an unlimited number of repeats. If the output is routed back to the front end at greater than unity gain, what would happen? Each repeat would grow in level until the processor’s output was completely saturated in a hellacious storm of distorted echo.

Does that remind you of anything?

Acoustical Circuits

This is where my previous sentence comes into play: “I had simply failed to take into account the entire (and I do mean the ENTIRE) signal path I was working on.” I had temporarily forgotten that the delay line I was using for my tests had not magically started to exist in a vacuum, somehow divorced from the acoustical circuit it was attached to. Quite the opposite was true. The feedback setting on the processor might have been set at “negative infinity,” but that did NOT mean that processor output couldn’t return to the input.

It’s just that the output wasn’t returning to the input by a path that was internal to the delay processor.

I’ve talked about acoustical, resonant circuits before. We get feedback in live-audio rigs because, rather like a delay FX unit, our output from the loudspeakers is acoustically routed back to our input microphones. As the level of this re-entrant signal rises towards being equal with the original input, the hottest parts of the signal begin to “smear” and “ring.” If the level of the re-entrant signal reaches “unity,” then the ringing becomes continuous until we do something to reduce the gain. If the returning signal goes beyond unity gain, we get runaway feedback.

This is not fundamentally different from our delay FX unit. The signal output from the PA or monitor speakers takes some non-zero amount of time to get back into the microphone, just like the feedback to the delay takes a non-zero amount of time to return. We’re just not used to thinking of the microphone loop in that way. We don’t consciously set a delay time on the audio re-entering the mic, and we don’t intentionally set an amount of signal that we want to re-enter the capsule – we would, of course, prefer that ZERO signal re-entered the capsule.

And the “delay time” through the mic-loudspeaker loop is just naturally imposed on us. We don’t dial up “x number of milliseconds” on a display, or anything. However long it takes audio to find its way back through the inputs is however long it takes.

Even so, feedback through our mics is basically the same creature as our “hellacious storm” of echoes through a delay processor. The mic just squeals, howls, and bellows because of differences in overall gain at different frequencies. Those frequencies continue to echo – usually, so quickly that we don’t discern individual repeats – while the other frequencies die down. That’s why the fighting of feedback so often involves equalization: If we can selectively reduce the gain of the frequencies that are ringing, we can get their “re-entry level” down to the point where they don’t noticeably ring anymore. The echoes decay so far and so fast that we don’t notice them, and we say that the system has stabilized.

All of this is yet another specific case where the patterns of audio behavior mirror and repeat themselves in places you might not expect.

As it turns out, you can put this to very powerful use.

The Application

As I discussed in “Transdimensional Noodle Baking,” we can do some very interesting things with audio when it comes to manipulating it in time. Making light “late” is a pretty unwieldy thing for people to do, but making audio late is almost trivial in comparison.

And making audio events late, or spreading them out in time, allows you to examine them more carefully.

Now, you might not associate careful examination with fighting feedback issues, but being able to slow things down is a big help when you’re trying to squeeze the maximum gain-before-feedback out of something like a a monitor rig. It’s an especially big help when you’re like me – that is to say, NOT an audio ninja.

What I mean by not being an audio ninja is that I’m really quite poor at identifying frequencies. Those guys who can hear a frequency start to smear a bit, and instantly know which fader to grab on their graphic EQ? That’s not me. As such, I hate graphic EQs and avoid putting them into systems whenever possible. I suppose that I could dive into some ear-training exercises, but I just can’t seem to be bothered. I have other things to do. As such, I have to replace ability with effort and technology.

Now, couple another issue with that. The other issue is that the traditional method of “ringing out” a PA or monitor rig really isn’t that great.

Don’t get me wrong! Your average ringout technique is certainly useful. It’s a LOT better than nothing. Even so, the method is flawed.

The problem with a traditional ringout procedure is that it doesn’t always simulate all the variables that contribute to feedback. You can ring out a mic on deck, walk up, check it, and feel pretty good…right up until the performer asks for “more me,” and you get a high-pitched squeal as you roll the gain up beyond where you had it. The reason you didn’t find that high-pitched squeal during the ringout was because you didn’t have a person with their face parked in front of the mic. Humans are good absorbers, but we’re also partially reflective. Stick a person in front of the mic, and a certain, somewhat greater portion of the monitor’s output gets deflected back into the capsule.

You can definitely test for this problem if you have an assistant, or a remote for the console, but what if you have neither of those things? What if you’ve got some other weird, phantom ring that’s definitely there, and definitely annoying, but hard to pin down? It might be too quiet to easily catch on a regular RTA (Real Time Analyzer), and you might not be able to accurately whistle or sing the tone while standing where you can easily read your RTA. Even if you can carry an RTA with you (if you have a smartphone, you can carry a basic analyzer with you everywhere – for free) you still might not be able to accurately whistle or sing the offending frequency.

But what if you could spread out the ringing into a series of discrete echoes? What if you could visually record and inspect those echoes? You’d have a very powerful tuning tool at your disposal.

The Implementation

I admit, I’m pretty lucky. Everything I need to implement this super-nifty feedback finding tool lives inside my mixing console. For other folks, there’s going to be more “doing” involved. Nevertheless, you really only need to add two key things to your audio setup to have access to all this:

1) A digital delay that can pass all audio frequencies equally, is capable of long (1 second or more) delays, and can be run with no internal feedback.

2) A spectrograph that will show you a range of 10 seconds or more, and will also show you the frequency under a cursor that you can move around to different points of interest.

A spectrograph is a type of audio analysis system that is specifically meant to show frequency magnitude over a certain amount of time. This is similar to “waterfall” plots that show spectral decay, but a spectrograph is probably much easier to read for this application.

The delay is inserted in the audio path of the microphone, in such a way that the only signal audible in the path is the output of the delay. The delay time should be set to somewhere around 1.5 to 2 seconds, long enough to speak a complete phrase into the mic. The output of the signal path is otherwise routed to the PA or monitors as normal, and the spectrograph is hooked up so that it can directly (that is, via an electrical connection) “listen” to the signal path you’re testing. The spectrograph should be set up so that ambient noise is too low to be visible on the analysis – otherwise, the output will be harder to interpret.

To start, you apply a “best guess” amount of gain to the mic pre and monitor sends. You’ll need to wait several seconds to see if the system starts to ring out of control, because the delay is making everything “late.” If the system does start to ring, the problem frequencies should be very obvious on the spectrograph. Adjust the appropriate EQs accordingly, or pull the gain back a bit.

With the spectrograph still running, walk up to the mic. Stick your face right up on the mic, and clearly but quickly say, “Check, test, one, two.” (“Check, test, one, two” is a phrase that covers most of the audible frequency spectrum, and has consonant sounds that rely on high-mid and high frequency reproduction to sound good.)


See, what you’re effectively doing is finding the “hot spots” in the sound that’s re-entrant to the microphone, and if you move away from the mic you change where those hot spots are. So…

Stay put and listen. The first thing you’ll hear is the actual, unadulterated signal that went through the microphone and got delivered through the loudspeaker. The repeats you will hear subsequently are what is making it back into the microphone and getting re-amplified. If you hear the repeats getting more and more “odd” and “peaky” sounding, that’s actually good – it means that you’re finding problem areas.

After the echoes have decayed mostly into silence, or are just repeating and repeating with no end in sight, walk back to your spectrograph and freeze the display. If everything is set up correctly, you should be able to to visually identify sounds that are repeating. The really nifty thing is that the problem areas will repeat more times than the non-problem areas. While other frequencies drop off into black (or whatever color is considered “below the scale” by your spectrograph) the ringy frequencies will still be visible.

You can now use the appropriate EQs to pull your problem frequencies down.

Keep iterating the procedure until you feel like you have a decent amount of monitor level. As much as possible, try to run the tests with gains and mix levels set as close to what they’ll be to the show as possible. Lots of open mics going to lots of different places will ring differently than a few mics only going to a single destination each.

Also, make sure to remember to disengage the delay, walk up on deck, and do a “sanity” check to make sure that everything you did was actually helpful.

If you’re having trouble visualizing this, here are some screenshots depicting one of my own trips through this process:

This spectrograph reading clearly shows some big problems in the low-mid area.

Some corrective EQ goes in, and I retest.

That’s better, but we’re not quite there.

More EQ.

That seems to have done the trick.

I can certainly recognize that this might be more involved than what some folks are prepared to do. I also have to acknowledge that this doesn’t work very well in a noisy environment.

Even so, turning feedback problems into a series of discrete, easily examined echoes has been quite a revelation for me. You might want to give it a try yourself.

Audio Processing In Graphical Terms

A guest post for Schwilly Family 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.

The “doing of things” to audio can seem pretty abstract, and so I decided to write a piece that uses pictures to demonstrate signal processing. Go on and have a look.