Tag Archives: Power

A Tiny Bit Of Practical Math For Audio Folks

If a number is part of a nonlinear operation, the only way to extract that number is through a nonlinear operation.

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.

logcurveWant to use this image for something else? Great! (No high-res on this one. Sorry – I forgot.)
I personally think it’s very handy for audio humans to be able to look at concepts quantitatively. That is, with measurements. It’s a great way to suss out what’s really happening with an audio rig.

Quite often, when trying to work with audio issues involving math, you end up with an unknown in an “inconvenient” place. Finding the unknown means some algebraic acrobatics – which wouldn’t be a big deal if not for the mathematics of sound being funky.

Audio math isn’t just bog-simple linear operations. It’s also nonlinear in nature, and the nonlinear bits (that is, logarithms) can make the algebra confusing. It’s confusing to the point that folks like me, who’ve been involved with audio for a good long while, can still go about something in entirely the wrong way.

But there’s one thing I finally realized. It’s one of those things that was probably explained to me ages ago, but didn’t “take” for some reason. It’s a realization that makes things much easier:

For the purposes of algebra, a logarithm encapsulates the connected number or expression that REPRESENTS a number. You can NOT extract the connected number or expression through linear means.

If you just said, “What?” then don’t worry. I can give you an example.

Let’s say that you’ve got an amplifier that can output a momentary, undistorted peak of 500 watts into a loudspeaker connected to one of the channels. What you’re curious about is a ballpark figure regarding the continuous power involved when you reach that peak. You figure that the crest factor of the signals sent to the amp (the ratio of peak to RMS voltage) is about 12 dB. Remembering your basic audio math, you work this up:

10 log10 x/500 = -12 dB

In other words, an unknown number of watts compared to the known peak power of 500 watts is -12 dB. (The decibel in this case is being referenced to 500 watts.)

Dividing both sides of the equation by 10 is appropriate, because that “10” on the left is engaged in the linear operation of multiplication. As such, the linear operation of division is the inverse. You end up with:

log10 x/500 = -1.2 dB

Now – it’s very tempting to try a linear operation to “move x” to a convenient spot. You might think that dividing by x gets you this (which becomes easy to work out on a calculator):

log10 1/500 = -1.2 dB/x
-2.6989 = -1.2 dB/x
-2.6989x = -1.2 dB
x = -0.444 watts

Nope. That can’t be right. For a start, there’s no such thing as negative power. For another, 10 dB down from 500 watts is 50 watts, and 3 dB down from that is 25 watts, so the number -0.444 isn’t even close. Even if you didn’t know that, plugging -0.444 into the original equation yields an answer that doesn’t agree with the original conditions:

10 log10 -0.444/500 = -12 dB
log10 -0.444/500 = -1.2 dB
[Calculator Returns: Invalid Input] ≠ -1.2 dB

Remember what I said: The logarithm is encapsulating the “x/500.” That is to say, x/500 is NOT two numbers in this case. It’s one number, represented by an expression, and we’re trying to take the logarithm of it. The only way to get the number “x/500” out into a place where you can use linear math is to reverse the logarithm. Here’s where we were before things went wrong:

log10 x/500 = -1.2 dB

The inverse of a logarithm is an exponent. The logarithm’s base is nothing more exotic than the base number that the exponent raises, and the exponent itself is whatever is on the other side of the equation.

10^-1.2 = x/500

NOW you can use linear math.

0.0630 = x/500
31.548 = x

Put that back into the original equation, and things work out perfectly.

10 log10 31.548/500 = -12 dB
log10 31.548/500 = -1.2 dB
log10 0.0630 = -1.2 dB
-1.2 dB = -1.2 dB

So, if you remember that extracting numbers from nonlinear operations requires an inverse nonlinear operation, you’ll figure out that the continuous power across your speakers is about 31 watts.

(Incidentally, this is one of the reasons why big PA systems are so big, but that’s a discussion for another day…)

The Calculus Of Music

There’s a lot of math behind the sound of a show, but you don’t have to work it out symbolically.

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.

This post is the fault of my high-school education, my dad, and Neil deGrasse Tyson.

In high-school, I was introduced to calculus. I wasn’t particularly interested in the drills and hairy algebra, but I did have an interest in the high-level concepts. I’ve kept my textbook around, and I will sometimes open it up and skim it.

My dad is a lover of cars, and that means he gets magazines about cars and car culture. Every so often, I’ll run across one and see what’s in the pages.

About a month ago, I was on another jaunt through my calculus book when I happened upon a car-mag with an article by Neil deGrasse Tyson. (You know Dr. Tyson. He’s the African-American superstar astrophysicist guy. He hosted and narrated the new version of “Cosmos.”) In that article was a one-line concept that very suddenly connected some dots in my head: Dr. Tyson pointed out that sustained speed isn’t all that exciting – rather, acceleration is where the fun is.



The rate of change.

Derivative calculus.

Exciting derivative calculus makes for exciting music.


Let me explain.

Δy/Δx: It’s Where The Fun Is!

The first thing to say here is that there’s no need to be frightened of those symbols in the section heading. The point of all this is not to say that everybody should reduce music to a set of equations. I’m not suggesting that folks should have to “solve” music in a symbolic way, as a math problem. What I am saying is that mathematical concepts of motion and change can be SUPER informative about the sound of a show. (Or a recording, too.)

I mean, gosh, motion and change. That sounds like it’s really important for an art form involving sine waves. And vibrating stuff, like guitar strings and loudspeakers and such.


Those symbols up there (Δy/Δx) reflect the core of what derivative calculus is concerned with. It’s the study of how fast things are changing. Δy is, conventionally, the change in the vertical-axis value, whereas Δx is the change in the horizontal-axis value. If you remember your geometry, you might recall that the slope of a line is “rise over run,” or “how much does the line go up or down in a given horizontal space?” Rise over run IS Δy/Δx. Derivative calculus is nothing more exotic than finding the slopes of lines, but the algebra does get a bit hairy because of people wanting to get the slopes of lines that are tangent to single, instantaneous points on a curve YOUR EYES ARE GLAZING OVER, I KNOW.

Let’s un-abstractify this. (Un-abstractify is totally a word. I just created it. Send me royalties.)

Remember that article I wrote about the importance of transients? Transients are where a change in volume is high, relative to the amount of time that passes. An uncompressed snare-drum note has a big peak that happens quickly. It’s the same for a kick-drum hit. The “thump” or “crack” happens fast, and decays in a hurry. The difference in sound-pressure from “silence” to the peak volume of the note is Δy, and the time that passes is Δx. Think about it – you’ve seen a waveform in an audio-editor, right? The waveform is a graph of audio intensity over time. The vertical axis (y) is the measure of how loud things are, and the horizontal axis (x) is how much time has passed. Like this:

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

For music to be really exciting, there has to be dramatic change. For music to be calming, the change has to be restrained. If you want something that’s danceable, or if you want something that has defined, powerful impact regardless of danceability, you’ve got to have room for “big Δy.” There has to be space for volume curves that have steep slopes. The derivative calculus has to be interesting, or all you’ll end up with is a steady-state drone (or crushingly deafening roar, depending on volume) that doesn’t take the audience on much of a ride. (Again, if you want a calming effect, then steady-state at low-volume is probably what you want.) This works across all kinds of timescales, by the way. Your music might not have sharp, high-speed transients that take place over a few milliseconds, but you can still move the audience with swells and decrescendos that develop over the span of minutes.

Oh, and that graphic at the top of the page? That’s actually a roughly-traced vocal waveform, with some tangent-lines drawn in to show the estimated derivatives at those points. The time represented is relatively small – about one second. Notice the separation between the “hills?” Notice how steep the hills are? It turns out that the vocal in that recording is highly intelligible, and I would strongly argue that a key component in that intelligibility is a high rate of change in the right places. Sharp transitions from sound to sound help to tell you where words begin and end. When it all runs together, what you’ve got is incoherent mumbling. (This even works for text. You can read this, because the whitespace between words creates sharp transitions from word to word. This,ontheotherhand…)

Oh, and from a technical standpoint, headroom is really important for delivering large “Δy” events. If the PA is running at close to full tilt, there’s no room to shove a pronounced peak through it all. If you want to reproduce sonic events involving large derivatives, you have to have a pretty healthy helping of unused power at your disposal.

Now, overall level does matter as well, which leads us into another aspect of calculus.

Integral Volume

Integral calculus contrasts with derivative calculus, in that integration’s concern is with how much area is under the curve. From the perspective of an audio-human, the integral of the “sonic-events curve” tells you a lot about how much power you’re really delivering to those loudspeaker voice-coils. Short peaks don’t do much in terms of heating up coil windings, so loudspeakers can tolerate rather high levels over the short term. Long-term power handling is much lower, because that’s where you can get things hot enough to melt.

From a performance perspective, integration has a lot to say about just how loud your show is perceived to be. I’ve been in the presence of bands that had tremendous “derivative calculus” punching power, and yet they didn’t overwhelm the audience with volume. It was all because the total area under the volume curve was well managed. The long-term level of the band was actually fairly low, which meant that people didn’t feel abused by the band’s sound.

This overall concept (which includes the whole discussion of derivatives) is a pretty touchy subject in live audio. That is, it can all be challenging to get right. It’s situationally dependent, and it has to be “just so.” Too much is a problem, and too little is a problem. For example, take this blank graph which represents a hypothetical, bar-like venue where the band hasn’t started yet:

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

If the band volume’s area under the curve is too small, they’ll be drowned out by the talking of the crowd. Go too high, though, and the crowd will bail out. It’s a balancing act, and one that isn’t easy to directly define with raw numbers. For instance, here’s an example of what (I think) some reggae bands might look like over the span of several seconds:

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

The “large Δy” events reach deep into the really-loud zone, but they’re very brief. Further, there are places where the noise floor peeks through significantly. This ability for the crowd to hear themselves talking helps to send the message that the band isn’t too loud. Overall, the area under the curve is probably halfway to three-quarters into the “comfortable volume” zone. Now, what about a “guitar” band:

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

The peaks don’t go up quite as far. In terms of sustained level, the band is probably also halfway to three-quarters into the comfortable zone – and yet some folks will feel like the band is a bit loud. It’s because the sustained roar of the guitars (and everything else) is enough to completely overwhelm the noise floor. The crowd can’t hear themselves talk, which sends the message that the band’s intensity is higher than it is in terms of “pure numbers.”

As an aside, this says a lot about the problems of the volume war. At some point, we started crushing all the exciting, flavorful, “large Δy” material in order to get maximum area under the curve…and eventually, we started to notice just how ridiculous things were sounding.

And then there’s one of my pet peeves, which is the indie-rock idiom of scrubbing away at a single-coil-pickup guitar’s strings with the amp’s tone controls set for “maximum clang.” It creates one of the most sustained, abrasive, yet otherwise boring noises that a person can have the displeasure of hearing. Let me tell you how I really feel…


Excitement, intelligibility, and appropriate volume levels are probably just a few of the things described by the calculus of music. I’ll bet there’s more out there to be discovered. We just have to keep our cross-disciplinary antennae extended.

Transient Impact

Music that hits hard requires careful management of the parts that don’t hit hard.

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.

A few weeks ago, I had the unexpected pleasure of working with a band called “Outside Infinity.” I say that the pleasure was unexpected because I had some major concerns going into the show. Metal, as a genre, can be pretty challenging in a small space. The sheer volume can be tough (or even impossible) to work with, and the arrangements are often quite dense – which compounds the volume problem. Several instruments banging away at full-blast can make for lots of challenges when trying to differentiate each part of a mix.

Outside Infinity had none of those problems. In fact, they were a prime example of how heavy metal – or any type of music that you want to “hit hard” – actually achieves that goal. (They were so much fun to listen to that I’m pretty sure I had a stupid grin on my face for large portions of the night.) I was really impressed by the sound that they had crafted, and I started to think about it.

Why were they so much fun?

Why did they capture what I’ve loved about heavy metal in the past?

Why did their sound have what so many rock and metal bands want, but so often fail to achieve?

I think that the generalized answer to all of those questions is this: Transient impact.

The Stopping Is As Important As The Starting

There are a number of necessary elements to a really great song performed live in a really great way. The lyrics have to be interesting, of course, and a memorable melody (or overall musical theme) is required. Skipping those steps will efficiently torpedo a tune’s ability to grab and hold an audience. There’s more, though: The overall sound of the song has to keep the listener interested. It’s analogous to eating a meal that leaves you remembering the food for years. Every bite is delicious, yes, but certain bites contain an extra explosion of flavor that plays on the mouth and tongue…and then dissipates. That “taste transient” pokes out from the “steady state deliciousness” of the rest of the meal, creating an ebb and flow of special delight, anticipation, and reward.

But if that burst of flavor just continued unabated, with no steady-state to contrast it against, then the “burst” wouldn’t be attention-getting anymore. It would BE the steady-state, and would quickly become unremarkable.

Sound behaves in a way that’s fundamentally the same. We perceive it differently, and the time-scales involved are sometimes much shorter, but the transient content is still the basis of what holds attention. Transient content is the determining factor behind the (ironically) nebulous idea of music that’s “really defined.” In music that aims to convey power and force, sounds that hit above the steady-state, and then swiftly decay are what cause the individual parts to “slam into you.” Everything just banging away at full throttle, continuously, for several minutes, has no impact. No spark of flavor. The brain starts to have trouble distinguishing the music from noise, because of the lack of anything to lock on to.

The mastery of stopping notes at the right time is what creates epic riffs. The mastery of creating a pleasing steady-state, which is then punctuated by sharp, sonic flavors, is the essence of the “thunderous” rock show.

…and because transients are all about proportionality, it is entirely possible to create a pile-driving artillery barrage of a show within the confines of a small venue. More on that later. First:

Dynamics And Articulation

Music, especially rock and metal, has a long history of breaking rules and pushing boundaries. This is what drives innovation, and it’s a good thing. However, there are certain rules that can’t truly be broken successfully. Those rules are the ones that are based in fundamentals of the physical universe and human perception.

One such rule is that, for a particular musical part to seem “big,” the other parts around it must be proportionally small. There are different ways of achieving this, but it all pretty much boils down to volume. The “small” part must either be quieter across the entire audible spectrum, or quieter across the most important part of the spectrum occupied by the “big” part. Especially in the small-venue context, plenty of bands shoot themselves in the foot with this. I’ve heard too many groups that interpret the instrumental breaks of their songs as “there’s no vocal, so now all the instruments should play as loudly as they can, occupy every frequency possible, and we’ll just hope that the audio-human can crank the actual solo above all that.”

(The best bands avoid this problem by interpreting the solo instrument as being “the new vocal,” and thus they keep all the other instruments in a supporting role until it’s their turn to be in front.)


In music, there are lots of broad-brush ways to accomplish this necessary contrast. There are the overall dynamics of individual parts across a number of beats, and there are also the rests – where a part is silent for a time. Whether formally or informally, these contrasts can be reliably notated. It’s pretty easy to explicitly define the necessary negative space, whether by a symbol for a rest, a “pp” for being very quiet, or a scribbled note saying, “For the fingerpicked guitar part, no drums at all and everybody else turns way down.”

There’s something else, though, that’s required for mastery. It’s hard to explicitly notate. It’s articulation.

Articulation (as I see it) is the manner in which notes and chords are played. It’s a crucial part of getting transients to contrast with the rest of the music, because it involves dynamics and rests that are too short and frequent to write down…and yet have a massive effect on how other parts sound. Playing a power chord with a “micro rest” at the end can be key to getting a kick-hit to punch through. Making that kick-hit decay into silence quickly can make room for a note from the bass. Going through a run of notes where each tone is connected, but there’s a very slight volume drop just before the next sound, can make for a clean and precise solo line. The singer hitting a big note and then backing off means that they can help support that solo line without a miniature volume war erupting.

The very best bands have a reliable handle on making this all work – even if they’re not explicitly aware of what they’re doing. Their riffs are powerful and defined because the individual notes have space around them. Their drum hits are forceful and satisfying because there’s space for them to stick up above everything else – and yet the drums don’t overpower the tonal instruments, because the individual hits decay into the “steady state volume” before the tonals hit THEIR next transient.

This leads me into that promised bit about how this is possible in small venues.

The SPL Difference Is The Key, Not The Absolute SPL Magnitude

A common mistake in trying to reproduce big-show impact in a small room is trying to replicate the big-show’s absolute SPL (Sound Pressure Level). It’s very easy to think that “so and so sounded huge, and they were making about 115 dBC in the center of the crowd, so that’s what we should do.” What tends to happen, though, is that reaching that kind of level chews up all the power available in a small-venue audio rig. The result is a show that doesn’t have those oh-so-cool transient hits, because there’s just no room for them to assert themselves.

Instead of defeating yourself with excessive volume, what you have to think about is WHY the big-show PA was making 115 dBC in the center of that huge crowd. It’s proportionality. Several thousand humans having a big party can make a surprising amount of noise – and so, to be clearly audible, the audio rig has to make even more noise. If a giant crowd is hollering at 105 dBC, then the audio-human running the system up to 115 dBC is understandable…if maybe a bit excessive. (Or not. It depends.)

From that previous paragraph, you can see that the proportionality between the steady-state volume of the crowd and the steady-state volume of the band was 10 dB. In certain kinds of small venues, that might be a little bit too much. A window of +6 to +9 dB of continuous level above the crowd is worth trying for in most contexts – in my opinion. Note that the “trying” part is most likely going to be in the downward direction. Getting loud is surprisingly easy, but holding your level in check to a point where the crowd is still pretty-darned audible is HARD. It’s hard for bands, and hard for audio-humans, but it’s worth trying for.

The point of holding your continuous level down, beyond just being nice to your crowd, is that it creates space for your show’s transients. Especially if you’re a metal band, and you want that big, thunderous kick, your best chance is to be had by giving the PA lots of room. If the audio-human has to run the system at full tilt just to keep up, then there probably won’t be enough power available to put those chest-thumping transients where you want them to go. On the other hand, keeping the show’s continuous level at a manageable point means that there’s reserve power – reserve power that has to be available to create large, proportional differences for things that need accenting.

Running the audio for Outside Infinity was fun, because they had an instinctive handle on negative space and transient impact. There was plenty of power available for the musical peaks, because the continuous level of the band was appropriate and comfortable. They knew how to articulate their notes so that the music was sharp and defined. I was really impressed.

And, if you take the time to think about your music’s transients, you’ll probably also have a good shot at being that impressive.

On Powered Speakers (And Other “Black Boxes”)

The commoditization of live-sound is enabled by manufacturers removing unknowns from their equations.

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.

When I talk about a “black box,” I’m not thinking of an aircraft’s flight recorder. I’m not even thinking of a device enclosure that’s black.

And seriously, as much as we say that there are a lot of ugly, black-colored boxes in live-sound, let’s be real. Most of them are really just a very deep gray. If they were actually black, they would absorb all light and completely disappear when they were in shadow. Like ninjas. Ninjas that amplify bands. (That would be a great movie.)

Okay, where was I?

When I say, “black box,” what I’m getting at is a concept. It’s the idea that the user of a device doesn’t know how the device works – or, they might now, but they aren’t required to know. Whether or not people are conscious of it, this is a central factor in the commoditization of technological devices. That is, for people to regard technological thingamabobs as “common, everyday” sorts of tools, those folks have to be in a world where understanding the internal functioning of the tool is not required.

A fine example of this is the personal computer. As the years have gone by, hardware and software manufacturers have progressively “black boxed” their offerings. In the computer’s infancy, operating a computer meant you had to have a lot of detailed knowledge about what the computer was doing. Nowadays – not so much. Almost everything is handled invisibly (which is great, until something breaks). Whether or not you think this is good or bad, this reality of “it just works” has allowed the personal computer to become a thoroughly mundane item. Having and using a computer isn’t a special thing anymore…in fact, it’s rather more surprising if someone DOESN’T have a computer that they use regularly.

In the same way, live-sound is also far more commoditized than it used to be. For instance, I’m betting that most readers of this site have never constructed a power amplifier. I know that I haven’t. Most of you probably haven’t built your own mixer. I know I haven’t.

But, in the early days, building your own gear from the ground up was often required. You couldn’t just head on over to the store and browse a vast selection of poweramps, loudspeakers, mixers, and whatever else. Before pro-audio (as we know it) really took hold as a market segment, the people pushing the boundaries were working by building things that either didn’t exist, or didn’t exist in enough quantity that they could be easily gotten “off the shelf.”

Now, pretty much every audio device you can think of is already in existence. You can go online and positively drown in a million iterations and manufacturer-specific takes on all manner of gear. Even if you’re thinking of something rather narrowly defined, like a 2-way active crossover, you won’t have any trouble finding a bunch of options to pick through.

It’s funny that I just mentioned active-crossovers, because it’s possible that you may never have to buy one. That’s because of one particular class of “black box” product: The powered loudspeaker.


The powered or “active” loudspeaker is hardly a monolithic sort of entity. They exist in all shapes and sizes, with some being vastly more capable than others. There are plenty of active loudspeakers that put on a facade of advanced engineering, but really aren’t much more complicated than you or I connecting a rackmounted power amp to a “full-range” loudspeaker. Even so, every powered loudspeaker on the planet shares a common trait:

They all encapsulate devices with diverse operations into a single, functional unit.

In other words, powered loudspeakers stick components with very different purposes into one box. In the most basic case, you have a power amplifier bundled up with a loudspeaker. The power amp takes a relatively small input voltage and delivers a corresponding, high-voltage, high-current signal to a load. The loudspeaker takes a high-voltage, high-current signal and transduces it into sound-pressure waves. Obviously, these two actions are complementary, but they’re also very different. Encapsulating the two actions reduces complexity for the user. Where they once had to manage and connect the amplifier and loudspeaker as separate units, they now only have to look after one unit and one signal connection.

What can be missed, though, is that this simplification by encapsulation involves a very profound “exchange.” This exchange puts tremendous capability in the hands of people who would not be able to access it otherwise.

Many Unknowns For The User, Almost No Unknowns For The Manufacturer

A non-encapsulated system is a pretty complex thing to build and deploy. Let’s take the case of a fully-processed, biamplified loudspeaker. (Biamplification is the use of independent amplifiers for low and high-frequency signals.) To construct and operate an un-encapsulated, fully-processed, biamped audio rig, the following has to happen:

  1. You have to pick out, purchase, rackmount, and connect some sort of equalizer.
  2. You have to do the same for an active, two-way crossover.
  3. You might also want some dynamic filters – or even full-fledged dynamic EQ – for each crossover output.
  4. For both crossover outputs, you will need to have a limiter. If you want to get fancy, you’ll need two limiters – one that can determine and limit the RMS level of a signal, and one that “brickwalls” peak levels.
  5. You’ll need an alignment delay for one channel or the other. (Alignment delay is fraction-of-a-millisecond control over when a signal arrives. Effect delay has much coarser control over the time involved, and it’s also mixed with the unmodified signal to create the sound of an echo.)
  6. You will need two channels of amplification. The power available from each channel will need to be more than what the drivers can handle. I’ll explain why in just a bit.
  7. Now you can add a cabinet with an LF and HF driver.

If you’ve got all that done, now you get to do a bit of science. First, you pre-configure the crossover based on recommendations from the loudspeaker manufacturer.

You next have to figure out what input voltages to the amplifiers correspond with output voltages that – just barely – won’t destroy your drivers. You set the peak-stop limiter accordingly, with the RMS-sensing limiter in place as a backup. The reason that you got a “too powerful” amp is that even VERY heavily limited signals usually end up having a continuous power that’s one quarter of the peaks. As such, getting the maximum, “sane,” real-world performance possible means using amps that can deliver more continuous power than the drivers are rated for…and then limiting the continuous power to something safe while letting some of the peaks through. (If you want to be really dangerous, you could set RMS limiter only. It will probably be a while before something gets destroyed. Maybe.)

By the way – if you end up trying any of this, and you blow something up, I am NOT liable. It’s your funeral, okay?

Now you have to find an environment that’s as anechoic as possible (or go outside), and set up a measurement rig. The first thing to do is figure out which driver’s sound arrives “late” when compared to the other. You then apply the alignment delay to the “early” driver, so that signals from both the HF and LF elements hit the listener at the same time. Next, you measure the whole thing and apply EQ to make the response as flat as possible. If you’re ambitious enough, you run up the system to full-throttle and note how the response changes. You can then set dynamic EQs to keep the response flat (or filter out damaging LF energy) at high levels.

Oh, and you can always try some different crossover slopes to see what has the best phase and amplitude response.

So, yeah. You could buy all that for hundreds or thousands of dollars, and spend all that time dialing it in (assuming that you know what you’re doing), or…

…you could live with all of the above being unknown to you, but known to the manufacturer. If you’re willing to do that, then for a few hundred bucks you can purchase a powered box. That powered box will have had that whole mess up there done for it already. You just plug it into the wall, put some signal into it, and off you go.

See, when all of those components are encapsulated by an equipment builder, there’s an exchange that’s basically inevitable. The inner workings of the system become an unknown for you, the user. In trade, the configuration of all those components is now intimately understood and highly optimized by the manufacturer. This creates an integrated, powerful, black-box system that you can just use, with minimal effort. This especially gets around some of the problems I discuss in Dirty Secrets About Power: Manufacturers don’t have to deal with as many unknowns regarding how their equipment will be used, and you don’t have to deal with semi-knowns about what amp to mate with what loudspeaker cabinet.

In closing, let me be clear. I advocate being curious. I’m in favor of knowing what’s happening inside your gear, at least to whatever extent is practicable. I’m all for building things, and doing experiments. I’ve got access to some gear that I want to rebuild, to see just how effectively I can do a “biamped, externally powered and processed” loudspeaker rig. At the same time, the reality is that black-box products have created a world where you can just plug something in and get decent (if not stellar) results.

Dirty Secrets About Power

The amount of power actually being delivered to your loudspeakers might not be what you think. What power IS getting delivered might not be doing what you think.

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.

I’m pretty sure that power – that is, energy delivered to loudspeaker drivers – is one of the most misunderstood topics in live-audio. It’s an area of the art that’s often presented in a simplified way for the sake of convenience. Convenience is hardly a bad thing, but simplifying a complex and mission-critical set of concepts can be troublesome. For one, misinformation (or just misinterpretation) starts to be viewed as fact. Going hand-in-hand with that is the phenomenon of folks who mean well, but make bad decisions. These bad decisions lead to the death of loudspeakers, over and under spending on amps and speakers, seemingly reckless system operation…the list goes on.

So, with all the potential problems that can be caused by the oversimplification of the topic “Powering Loudspeakers,” why does “reduction for the sake of convenience” continue to occur?

I think the answer to that is ironically simple: The proper powering of loudspeakers is, in truth, maddeningly complex. There are lots of “microfactors” involved that are quite simple, but when they all get stuck together…things get hairy. At some point, educators with limited time, equipment manufacturers with limited space in instruction manuals, and established pros with limited patience have to decide on what to gloss over. (I’ve done it myself. Certain parts of my article on clipping let some intricacies go without complete explanation.)

With that being the case, this article can’t possibly cover every little counter-intuitive detail. What it can do, however, is give you some idea of how many more particulars are actually out there, while also giving you some insight into a few of those particulars.

So, in no particular order…

Dirty Secret #1: Amp And Speaker Manufacturers Assume A Lot

You may have heard the phrase “Assume Nothing.” That saying does NOT apply to the people who build mass-produced loudspeakers and amplifiers. It doesn’t apply because it CAN NOT apply – otherwise, they’d never get anything built, or their instruction manuals would be gigantic.

Amplifier manufacturers, on their part, assume that you’re going to use their product with mostly “musical” signals. They also assume that you can put together a sane system with the “how to make this thing work” information they provide in their documentation. Further, they make suggestions about using amplifiers with continuous power ratings that are greater than the continuous power ratings of your speakers, because they assume that you’re not going to drive the amp up to its clip lights all the time.

Loudspeaker manufacturers also assume that you’re going to drive their boxes with music. They also ship products with the assumption that you’ll use the speaker in accordance with the instructions. They publish power ratings that are contingent on you being sane, especially with your system equalizers.

The upshot of it all is that the folks who make your gear also make VERY powerful assumptions about your ability to use their products within the design limits. They do this (and disclaim a lot of responsibility), because a ton of factors related to actual system use have traditionally been outside their control. Anytime you read an instruction manual – especially the specifications page – take care to remember that the numbers you see are simplifications and averages that reflect a mountain of assumptions.

Dirty Secret #2: Musical Signals Don’t Get You Your Continuous Power Rating

The reason that technical folks distinguish between signals like sine waves, pink noise, and “music” is because they have very different power densities. Sine waves, for instance, have a continuous level that’s 3 dB below their peak level. Pink noise often has to have an accompanying specification of “crest factor” (the ratio between the peak and average level), because different noise generators can give you different results. Some pink noise generators give you a signal with 6 dB between the peak and average levels. Others might give you 12 dB.

Music is all over the map.

Some music signals have peaks that are 20+ dB above the average power. Of course, in our current age of “compress and limit everything,” it’s common to see ratios that are much smaller. I myself use rather aggressive limiting, because I need to keep a pretty tight rein on how loud the PA system can go. Even so, my peak levels tend to be about 10 dB above the average level.

So if you’ve got an amp that’s rated for “x” continuous watts, and you drive the unit all the way to its undistorted peak, music is probably giving you x/10 watts…or less. In my case, the brickwall limit that I set is usually 10 dB below clip, which means that my actual continuous power is something like 5 watts per channel. This calculation is pretty consistent with what I think the speakers are actually doing, because they get about 96 dB @ 1 watt @ 1 meter. Five watts continuous would mean about 103 dB SPL per full-range box, and there are two full-range boxes in the PA, so that’s 106 dB total…yup, that seems about right.

Yeah, so, your system? If you’re driving it with actual music that isn’t insanely limited, you can go ahead and divide your amp’s continuous power rating by about 10. Don’t get overconfident, though, because you can still wreck your drivers. It’s all because…

Dirty Secret #3: Power Isn’t Always Evenly Distributed

Remember that bit up there about manufacturers making assumptions? Think about this sentence: “They publish power ratings that are contingent on you being sane, especially with your system equalizers.”

Dirty secret #2 may have you feeling pretty safe. In fact, you may be thinking that secret #2 directly contravenes some of the things that I said about cooking your loudspeakers with an amp that’s too big.

Hold up there, chum!

When a loudspeaker builder says that the system will handle, say, 500 watts, what they actually mean is: “This system will survive 500 watts of continuous input, as long as the input is distributed with roughly equal power per octave.” Not everything in the box will take 500 watts without dying. In particular, the HF driver may be rated for a tenth – or less – of what the total system is advertised to do. Now, if you combine that with a system operator who just loves to emphasize high-frequency material (“I love that top-end snap and sizzle, dude!”), you may just be delivering a LOT of juice to a rather fragile component…

…especially if the operator uses a huge amp, because they’re under the false impression that amp headroom = safety. A 1000 watt amplifier, combined with a tech who drives hard, scoops the mids, and has boxes with passive crossovers, is plenty capable of beating a 50-watt-rated HF driver into the ground.

On the flipside, a system without protective filtering on the low-frequency side can get killed in a similar way. Some audio-humans just HAVE to “gun”the low-frequency bands on their system EQ, because “boom and thump are what get the girls dancing, dude!” Well, that’s all fine and good, but most live-sound speakers that are reasonably affordable can’t handle deep bass at high power. Heck, the box that the drivers are in often acts as a filter for material that’s below about 40 Hz.

Of course, there may not be an electronic filter to keep 40 Hz and below out of the amplifier, or out of the LF driver. Thus, our system operator might just be dumping a huge amount of energy into a woofer without actually being able to hear it. The power doesn’t just disappear, of course, which means that “driver failure because of too much power at too low a frequency” might be just around the corner.

Dirty Secret #4: Accidents Aren’t Usually Musical Signals

Building on what I’ve said above, I should be clear that folks do get away with using overpowered amps (for a time) because of feeding them “music.” They end up keeping the peaks at a reasonable level, and so the continuous power stays in a safe place as well.

Then, something goes wrong.

Maybe some feedback gets really out of control. Maybe somebody drops a microphone. All of a sudden, you might have a high-frequency sine-wave with peaks – and continuous level – that’s far beyond what a horn driver can live with. In the blink of an eye, you might have a low-frequency peak that can rip a subwoofer cone.


Dirty Secret #5: Squeezing Every Drop Of Performance From Something Is For Either Amateurs Or Rich People

This secret connects pretty directly with #3 and #4. Lots of folks worry about getting every single dollar’s worth of output from a live-audio rig. It’s very understandable, and also very unhealthy. To extract every possible ounce of output from a loudspeaker system requires powerful, expensive amplifiers that have the capability to flat-out murder the speakers. For this reason, “performance enthusiasts” are either people who can’t afford to buy both more power AND more speakers, or they’re people who can afford to buy (and fix, and fix, and fix again) a lot of gear that’s run very hard.

The moral of the story is that your expectation needs to be that – in line with secret #3 – getting continuous output consistent with about 1/10th of a rig’s rated power is actually getting your money’s worth. If you don’t have enough acoustical output at that level, then you either need to upgrade to a system that gets louder with the same number of boxes, or you need to buy more loudspeakers and more amps to expand your system.

Dirty Secret #6: More Power Means More Than Just Buying More Amps

This follows along with secret #5. If you want more power, then you need more gear. That seems simple enough, but I’m convinced that linear PA growth is accompanied by geometric “support” growth.

What I mean by this is that getting ahold of a more powerful PA is more than just getting the amps and speakers together. More power means heavier and more expensive amp racks, or more (and more expensive because of quantity) amp racks. It may mean that you have to construct patch panels to keep everything organized. More PA power also means that you need more AC power “from the wall” in the venue. Past a certain point, you have to start thinking about an actual power distro system – and that can be a major project with huge pitfalls in and of itself. You need more space for storage. You need a bigger vehicle, if you’re going to transport it all.

Getting more power doesn’t just mean more of the “core” gear that creates and uses that power. It means more of everything that’s connected to that gear.

Dirty Secret #7: The Point Of Diminishing Returns Occurs Very Quickly. Immediately, In Fact.

The last secret is also, in some ways, the biggest bummer. Audio is a logarithmic affair, which means that the gains you get from spending more money and providing more power to a system begin decreasing as soon as you even get started. I’m dead serious.

For example, let’s say you’ve got a loudspeaker that averages about 95 dB SPL @ 1 watt @ 1 meter. You put one continuous watt – one measly watt – across the box, and stand roughly three feet away. That 95 dB SPL seems pretty good. Now, you go up to two watts. Did you get 95 dB more? Nope – that would mean that you could get “space shuttle takeoff” levels out of one loudspeaker. Not gonna happen.

So…did you get 20 dB more?


10 dB?


You doubled the power, and got three decibels more level out of the speaker. That’s just enough of a difference to definitively notice that things have gotten louder. If you want three more dB, you’ll have to double the power again. So far we’re only at four watts, but I think you can see just how fast the battle for more output starts to go against you. If your system is running at full tilt, and you want more output, you’re going to have to find a way to “double” the system – and even when you do, you’ll only get a little more out of it. If you want to get 10 times as loud, you need 10 times as much total PA.

The vast majority of a PA system’s output comes from the first watt going into each box. It’s a fact that’s in plain sight, but it (and its ramifications) often aren’t talked about very much.

That makes it one of the dirtiest secrets of all.

Holistic Headroom

If you have zero headroom anywhere, you have zero headroom everywhere.

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.

“Headroom” is a beloved buzzword for audio craftspersons. Part of the reason it’s beloved is because you can blame your problems on the lack of it:

“I hate those mic pres. They don’t have enough headroom.”

“I’m always running out of headroom on that console’s mix buses.”

“I need to buy a more powerful amplifier for my subs, because I this one doesn’t have enough headroom.”

(I’m kinda tipping my hand a bit with that last one, in terms of this post being sort of a “follow on” to my article about clipping.)

Headroom is sometimes treated as a nebulous sort of concept – a hazy property that really good gear has enough of, and not-so-good gear doesn’t possess in the required quantity. In my opinion, though, headroom is pretty easy to define, and its seeming mysteriousness is due to it being used as a “blamecatcher” for things that didn’t go as planned.

Headroom, as I was taught, is “the difference between the maximum attainable level and the nominal level.” In other words, if a device can pass a signal of greater intensity than is required for a certain situation, then the device has some non-zero amount of headroom. For example, if your application requires a console’s main bus to pass 0 dBu (decibels referenced to 0.775 volts, RMS), and the console can pass +24 dBu, then you have 24 dB of headroom in the console.

(If it’s available, and ya ain’t usin’ it, it’s headroom.)

The overall concept is pretty easy to understand, but what a good number of folks aren’t taught, and often fail to realize for a good long while (this includes me), is that headroom is holistic, and “lowest common denominator.” That is to say:

Two or more audio components – whether electrical or acoustical – connected together all have the SAME effective headroom, and that effective headroom is equal to the LOWEST amount of headroom available at any point in the signal chain.

So…what the heck does that mean?

Everything Has A Maximum Level – Everything

To start with, it’s important to point out that hyphenated bit in the above definition. Especially because this is a site about live-performance, what you have to realize is that absolutely everything connected to that live performance has a maximum amount of appropriate signal intensity. Even acoustical sources and your audience qualify for this. Think about it:

A singer can’t sing any louder than they can sing.

A mic can only handle so much SPL.

A preamp can only swing a limited amount of voltage at its outputs.

Different parts of a console’s internal signal path have limits on how much signal they can handle.

A power amplifier can’t deliver an infinite amount of voltage.

Speakers handle a limited amount of power.

The people listening to the show have a finite tolerance for sound pressure.

…and every single one of these “components” is connected to the others. Sure, the connection may not be a direct, electrical hookup, but the influences of other parts of the system are still felt. If your system can create a “full tilt boogie” sound pressure level of 125 dB SPL C, but your audience will only tolerate about 105, then that lower level becomes your “don’t exceed” point. Go beyond it, and you effectively “clip” the audience…which makes your 20 dB of unused PA capability partially irrelevant. That leads to my next point.

Your Minimum Actual Headroom Is All You Effectively Have

Sometimes, a singer will “run out of gas.” They may have strained themselves, or they might not be feeling well, or they might just be tired. As a result, their maximum acoustical output drops by some amount.

Here’s the thing.

The entire system’s EFFECTIVE headroom has just dropped by that amount. If the singer is 10 dB quieter than they used to be, you’ve just lost 10 dB of effective headroom.

Now – before you start getting bent out of shape, complaining that your console’s mix bus headroom hasn’t magically changed, look at that paragraph again. The key is the word “effective.”

Of course your console can still pass its maximum signal. Of course your loudspeakers still handle the same power as they did a moment ago. As isolated components, their absolute headroom has not changed in any way.

But components working in a complete electro-acoustical system are not isolated, and are therefore limited by each other in various ways.

In the case of a singer getting worn out, their vocal “signal” drops closer to the noisefloor of the band playing around them. Now, if we were talking about an electrical device, the noisefloor staying the same with a decrease in maximum level above that noisefloor would be – what? Yes: A loss of headroom.

The way this affects everything else is that you now have to drive the vocal harder to get a similar mix. (It’s not the same mix, because there’s less acoustical separation between the singer and the band at the point of the mic capsule, but that’s a different discussion.) Because the singer’s overall level has dropped, your gain change might not be pushing you any closer to clipping an electrical device…but you are definitely closer to the point where your system will “ring” with feedback. A system in feedback, effectively, has reached its maximum available output.

Your effective headroom has dropped.

A Bigger Power Amp Isn’t Enough

Okay – here’s the bit that’s directly related to my “clipping” article.

The concept of holistic headroom is one of the larger and fiercer bugaboos to be found in the piecing together of live-audio rigs. As many bugaboos do, it grows to a fearsome size by feeding on misconceptions and mythology. There is a particular sub-species of this creature that’s both common and venomous: The idea that a system headroom problem can be fixed by purchasing more powerful amplifiers.

Now, if you’re constantly clipping your amps because the system won’t get loud enough for your application, then yes, you need to do something about the problem. However, what you need to do has to be effective on the whole, and not just for one isolated part of the signal chain. Buying a bigger amplifier will probably get you some headroom at the amplifier, but it might not actually get you any more effective headroom (which is what actually matters). If your old amplifier’s maximum level was equal to your speakers’ power handling, and the new amplifier is more powerful than the old one, then you’ve done nothing in terms of effective headroom.

The loudspeakers were already hitting their maximum level. As such, they had zero headroom, and your new amp is thus effectively limited to zero additional headroom. Your enormously powerful amp is doing virtually nothing for you, except for letting you hit your unchanged maximum level without seeing clip lights.

To be fair, the system will get somewhat louder because loudspeakers don’t “brickwall” at their maximum input levels. Also, the nature of most music is that the peaks are significantly higher than the continuous level, which lets you get away with a too-big amp for a while. You will get some more level for a while, but your speakers will die much sooner than they should – and when they do, your system will become rather quieter…


The point is that, if you want a system headroom increase of “x” decibels, then you have to be sure that every part of your system – not just one piece – has “x” more decibels to give you. If you’re going to get more power, you have to make sure that you also have that much more “speaker” to receive that power. (And this gets into all kinds of funny business, like whether or not you can buy speakers that are just as efficient as what you’ve had while handling more power, or whether you need to buy more of the same speakers, and if that’s a good idea because of arrayability, or…)

There’s also the question of whether or not a more powerful system is what your audience even wants. It all ties together, because headroom is holistic.

Clipping Does NOT Kill Loudspeakers

Clipping can be associated with a loudspeaker being cooked, but it isn’t really the cause.

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.

It’s very likely that – if you’ve ever been involved in live-sound – you’ve heard something like this: “You have to use very powerful amps on your speakers, because clipping will blow the drivers.”

This idea is one of the most long-standing live-sound myths in existence. Its ability to stubbornly persist as an accepted notion is remarkable, although not astounding. That is, it’s not surprising that the myth survives, because it’s backed up by observations made by intelligent people.

The problem is that the observations are being interpreted incorrectly. If you want to “have the right of it,” then you need to remember this:

Clipping is not a cause of speaker failure. It can occur alongside conditions that cause speaker failure, and it can precipitate conditions that cause speaker failure, but it is not actually dangerous in itself.

Okay. Fine. Where’s any support that what I just said is correct?

Well, to start with…

Thousands Of Guitar Amps Are – Miraculously – Still Alive

How many guitar players do you know that have to regularly replace the speakers in their rigs, because those speakers are constantly getting cooked? I don’t know any.

How many guitar players do you know who’s tone includes “crunch,” or “drive,” or “fuzz,” or “distortion,” or who love it when the amp “breaks up” or “saturates?” The number is probably close to “all of them.”


Distortion/ fuzz/ overdrive/ breakup/ whatever are ALL clipping. All of them. 100%. Some of them are clipping that happens in a circuit in front of the amp, which then gets passed through the amplifier “cleanly” (or not). Some of the most adored and sought-after sounds are the result of clipping the actual power-amp section – the bit that turns the signal into something with sufficient voltage and current to move loudspeakers around.

“Oh, but Danny, a distorted guitar amp is different from a distorted PA.”

Nope. Not in a fundamental sense.

Now, sure, an actually-clipping solid-state power amplifier may generate a different distribution of harmonics than an actually-clipping tube driven guitar amplifier, but the same thing is going on. A clipped signal is being pushed across loudspeaker drivers. Amazingly, the guitar amp’s speakers aren’t dying. Why?

Because the power they’re receiving is within their design limits. The distortion involved is barely relevant.

The Problem Is Too Much Power

What I’ve come to understand over the years is that, assuming everything else is copacetic, loudspeakers are only killed by amplifiers that supply too much power. It might be too much power for too long, or it might be too much power for a short time…at a low frequency.

That’s it.

So, if all kinds of guitar amplifiers aren’t killing their speakers, and if the problem is too much power, why does the myth persist? Why do people insist on mating big amplifiers to speakers, with the assumption that “headroom” will prevent drivers from meeting an untimely end? It’s pretty simple, actually – people tend to associate correlation with causation, even when the association is wrong.

Classic examples of this are found in human history. Something bad happens to a group of people at around the same time of a lunar eclipse, or when a comet is visible in the sky, and they start assuming that the cosmological event is the cause of their problem. In the same way, enough speakers have been wrecked while a clip light was illuminated to make people think that clipping was what wrecked their drivers.

…and so, they start to believe that running a really powerful amp without clipping is safer than running a less-powerful amp into clipping. It isn’t. Their original problem was that their “too small” amplifier can actually deliver a lot more power than they expect.

Amps Are More Powerful Than You Think. For An Instant.

Let’s say that we’ve just bought an amplifier, and we’re doing what we should be doing: We’re reading the manual. We get to the end, where the specifications live. The manufacturer says that the amp can deliver 400 watts per channel, continuous, at some impressively small distortion factor (like 0.02%), into an 8 ohm load.

Why is all that qualification necessary? It’s a 400 watt amplifier, right?

Not really.

As I’ve come to understand them, amplifiers are devices that put voltage across an attached device. You know – a speaker. Because they put voltage across the speaker, current flows. Because voltage and current are flowing, the circuit has an attendant amount of power being dissipated by the speaker – the power is converted to heat and sound.

The thing is, the voltage coming out of the amplifier is NOT constant. It’s not direct current…it can’t be. Direct current doesn’t change over time, so it can’t represent a sonic event. A sonic event changes over time by nature. No, the signal coming out of the amplifier is time variant. It’s alternating current, rather similar to what comes out of a “mains power” wall socket in a building. the primary differences are that the voltage coming out of the amplifier is significantly lower, and that we expect the signal from the amplifier to have a lot of frequencies present at similar voltages.

Music, in other words.

This creates a bit of a bugaboo. If the voltage from the amplifier varies as time passes, then the power delivered to the loudspeaker also varies as time passes. If we hook up a sine-wave generator to the amp, and then graph the amp’s output, we would get something like this:

There’s something very curious here. At the instant that the voltage is 0, no power is being presented to the loudspeaker. At that moment, we have a 0 watt amplifier. No voltage means no current, which means no power. Of course, at the very next instant the voltage is some non-zero value, which means that the power across the speaker is also non-zero.

What’s also curious – and key to this whole article – is what happens at the signal peaks. You’ll notice that they occur at 80 volts. If power is v^2/r (the square of voltage over the load resistance), then, for an instant, the amplifier delivers 800 watts to the speaker.

But it’s a 400 watt amp! What gives?

Remember all that “qualification” that was attached to that 400-watt number? It’s all required because the amp spends most of its time delivering more or less than 400 watts to our 8 ohm loudspeaker. The 400 watt figure is an average meant to convey what the amplifier can meaningfully do over the course of time, ultimately in terms of heat and sound produced by the speaker. For audio, we tend to find that values derived from RMS (Root Mean Square) voltages track well with how humans hear, so it’s very likely that the “continuous power” rating on our amp is the energy delivered from the RMS voltage that we can swing from the outputs.

For an amp that has a peak output voltage of +/- 80 volts, the sine-wave RMS voltage is about 56.57 volts. Using v^2/r, that comes out to 400 watts. If the loudspeaker is rated for 400 watts of continuous power, then we’re fine.

As long as we don’t push the amplifier too hard.

Not because of clipping, but because our amp is more powerful than we realize.

The Area Under The Curve

Here’s our diagram again. We’ve got ourselves a nice, undistorted signal. To help visualize the power being delivered to the loudspeaker, I’m going to fill in the “area under the curve,” or the space between 0 voltage and the amp’s output.

So, what happens if we push the amp beyond what it can do with inaudible distortion? Well, the amp can’t give us more voltage than it’s built to create, but it can give us the maximum voltage for a longer time. It might be able to do this “nicely,” by using internal dynamics processing to prevent the signal from actually generating a lot of nasty harmonics, or the amp might get into actual, unpleasant, super-saturated harmonic distortion overdrive. In either case, the output signal peaks flatten – and as they do, the continuous power delivered to the speaker gets closer and closer to the maximum power available from the amplifier. If I overlay the most extreme case of this over our original sine-wave, you can “see” the problem:

The closer you get to driving the amp into square-wave territory, the more that the RMS voltage and the peak voltage become the same thing. Assuming that the amp doesn’t go into thermal shutdown or engage other protection, you can deliver a LOT of continuous power into your loudspeaker. In terms of the example I’ve been using so far, you could be putting up to TWICE the loudspeaker’s rated power into the poor thing.

Do that for long enough, and the voice coil (or even something else) will overheat and fail. You’re left with smoke and silence.

Picking Up The Pieces

As you can see, the problem really isn’t clipping. Sure, clipping was involved in the process of wrecking the example loudspeaker, because it’s what had to happen for us to push our fictional amp into “too much power” territory.

What if we’d have used a bigger amp, though?

Here’s where things get into human psychology.

If we were willing to push a small amp into audible clipping (or even just limiting) for long enough to kill a loudspeaker, why would we think that we wouldn’t push a larger amp just as hard – if not harder? The big amp’s signal will stay nice and clean for much longer, and we might not be able to recognize the sounds of the drivers being beaten up. That being the case, we push our much larger amp well into the same overall power output, and our drivers start to get cooked again. Of course, we don’t see any clip lights, so we feel safe. The loudspeakers don’t die right away, because overpowering is rarely an “instant death” event, but they will die eventually. We didn’t see those evil little clip lights, though, so we assume that it’s just “wear and tear” or “defective drivers,” or “cheap gear.”

…but it was the same thing all along. Too much power.

Too much power is still the operative problem, even when true clipping at the amp hits a passive crossover and dumps extra energy into a high-frequency driver. Sure, if the amp hadn’t clipped, then that extra power wouldn’t have been present…but why were you running the rig so hard (and with such overpowered amps) that the power generated from the harmonics in a clipped signal could liquefy the HF driver’s voice coil? How could you stand to even listen to that? A smaller amp, clipped to the same degree, wouldn’t have killed the driver, although it would still have sounded terrible.

“Underpowering” isn’t the problem. Clipping isn’t the problem. Too much power and too much human error are the problem.

For The Love Of Mid

The material that’s critical for a mix is between about 200 Hz and 4000 Hz.

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.

We’ve all seen and heard it, in some way. You know what I mean. The “smiley face” EQ. “Scoop” switches. The midrange all the way down – and, optionally, the bass and treble CRANKED.


“Bedroom tone.”

Heck, most of us have been practitioners of this very thing. When trying to make something sound impressive, polished, and big, ruthlessly carving out the midrange is like the Dark Side of The Force: Quick, easy, and seductive.

Also, really bad for you in the end.

What a mix (live, studio, monitors, stage-volume, anything) actually stands or falls on is the midrange. Sure, you want the top and bottom octave to be in the right place, but they really aren’t as critical as you may have been led to believe.

So, why do people de-emphasize the midrange so much?

Tough, Lonely, Unexciting Rooms

There are all kinds of contexts that drive scooped, sizzle-thump tones. Getting into every detail could make for a very long, barely readable article. I think that you can get a decent picture by generalizing, though:

Midrange is common, unexciting, and – due to its criticality – annoying when it’s wrong.

See, humans hear midrange better than almost anything else. We’re great at detecting and analyzing human speech, because our lives basically depend on it. Human speech is all about midrange, and expressive, detailed vocalization is one of the things that makes humans actually…you know…human. We grow up hearing midrange. We communicate using midrange. We hear midrange all the time, in every possible place, in all kinds of contexts.

Midrange? More like, mundane-range.

When we come across a sound-generating item that can do the bits of the audible spectrum that are outside the boring and everyday, we fall in love pretty fast. “Bass” and “air” are like candy to our common meal of mid. They’re impressive. Fun. Exciting. Everything that those pokey, old-hat mids aren’t.

So, there’s a strong temptation to emphasize the fun bits at the expense of the boring parts.

At the same time, our particular human genius for detecting problems and unnatural weirdness in the mids makes us intolerant. Our brains are also VERY good at synthesizing missing information, especially when a lot of the basic cues are still intact. If your stereo or amplified instrument are in a not-so-acoustically-nice room, a quick fix is to yank out as much of the troublesome midrange as you can. The music still sounds fine, because the mids are still audible enough for you to imagine whatever you’re missing as you revel in the sounds that are emphasized.

The success of this is further enhanced by being alone, which is what leads to “bedroom sound.” With nothing else “in the mix,” you can hear your instrument just fine – and it sounds GREAT! All the midrange problems are sucked out, and the impressive “body” and “top” ends are dialed way up.

Awesome sauce.

Until real-life intervenes, of course.

Midrange Makes Mixes Musical

In the context of modern music, especially in small venues, what you have is an assemblage of amplified sounds that coexist with a lot of acoustical goings-on. For example, take a typical rock band’s rehearsal space. You’re probably going to run into an un-miced drumkit, one or two guitar amps, and a bass rig. The guitar and bass players, through electronics, have very immediate and dramatic control over the timbre of their instruments. Within the limits of their instruments and amplifiers, they can dial up some wild and weird tones.

On the other hand, the drummer can’t go quite as crazy. Sure, there’s a lot of variation to be had from shellpack to shellpack, especially with different heads, tunings, sticks, and everything else, but the reality is that most acoustic drumkits impart a tremendous amount of midrange into the room. If nobody else has much midrange left over, then the kit is going to obliterate the tonal parts of the song arrangements…unless, of course, the guitar and bass rigs are much louder than the drums.

So, here’s the major thing:

Sufficient midrange content is the primary and essential component of a tonal instrument’s place in a mix.

The reality is that, for all the excitement and fun that low and high-frequency information give us, there is very little actual music that occurs far below 200 Hz, or far above 4 kHz. It’s not that there isn’t ANY musical information beyond those areas – of course there is – it’s just that it usually isn’t critical to the actual song.

(Yes, bass guitars produce lots of fundamentals that are around or below 100 Hz, but the reality is that we mostly end up listening to the harmonic content of what the bassist is doing. Seriously – find yourself some songs with prominent, melodic basslines. Load the files into a DAW and filter everything below 200 Hz. I’ll bet that you can still hear the bass-human doing their thing.)

If the midrange content of a given part is de-emphasized in a big way, there is a very good chance that the part will disappear in an ensemble context. The flipside is that allowing everybody to have their own piece of the mids means that you’re much likely to get a better mix…especially when you’re playing live in a small room, where the interplay between purely acoustical sounds and amplified tones can be either beautiful or horrific.

Practical Considerations

The biggest take-away from this is that everybody – guitar players, bassists, vocalists, monitor guys, FOH (Front Of House) humans, and anybody else that I’ve missed – should resist the urge to “kill the mids.”

I should know, because I’ve had my own “scooping” bite me. Killed-mid vocals sound great in FOH and monitor world, right up until they have to be matched up with an actual band. At that point, you have to get the vocals VERY loud to get audible lyrics, and that can lead harshness, feedback, and an audience that wants to not be in the seats anymore.

I once had vocals dialed up in the monitors that sounded “super-studio.” Very hi-fi. It would have been great, except that when the band actually started playing you could barely hear the vocals in the wedges. You’ve gotta let those boxes “bark” a little if people are going to hear themselves sing.

On the flipside, I once worked with a band where one of the guitar players had a serious fascination with HF content. Once the drummer was playing, all you could hear out of that guitar was basically “eeeeeeessshhhh.” He would play these super-fast solos, but you couldn’t hear what he was doing. His actual notes were dialed out so far that, even when he was painfully loud and clearly in front of everybody else’s volume, you still only had a sort of screechy, clicky hiss to listen to.

There’s even a “technologic-economic” side to the whole thing. Making lots of low end and high end are tough things to do with an amplifier or a PA system. Killing the midrange and cranking the ends means that you’re probably wasting a ton of internal headroom and power-stage output on material that might not even be audible. If you want that material to be audible, you need lots of power and lots of speakers – and that’s spendy. Want to get the most out of more affordable gear? Get the midrange in the right place as the first step, and then use what you’ve got left over for the top and bottom.

The mids can be tough to love at first, but it’s a worthwhile relationship.

Buy A Little Amp

Large, powerful amplifiers were necessary in the early days of rock and roll. Not anymore.

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.

Nothing screams “Rock Band” like lots of gear. I myself will readily admit it: I LOVE the look of big, “ugly,” powerful, solidly-built amps and speaker cabs. You get all of that into a room, and by gum, everyone knows that you mean business.

Having people think that you mean business is a really good feeling. Heck, it’s an addictive feeling.

But it’s just a feeling.

What counts a lot more than looking like you mean business is to actually mean business, and then prove it beyond all doubt with your actual music. Proving you mean business in the small-venue context doesn’t require a lot of gear. It simply requires that you have enough gear.

Sure, you do want a bit of “cushion” or headroom, but a whole ton of it isn’t necessary. In fact, it can even be detrimental. We’ll get into that in a bit – but first, let’s talk about where the “big gear” thing came from.

The Days Before PA (As We Know It)

Way back when, in the days when men were real men, women were real women, and cars cost about as much as five tanks of gas today, you could count on one general rule for live-sound reinforcement:

You either made enough noise acoustically, or you had a dedicated amp.

The exception to this (but not by much) was the vocalists. Each vocalist might not have had their own PA, but the typical reality was that the PA only had a handful of inputs – and the PA only did vocals. The idea that you would put all the instruments through one sound rig was a foreign concept.

As a result, if you were doing a big show, you needed big amps. The drums might carry pretty well, but if you were going to get that guitar solo all the way to the back row, you needed serious firepower. Even as PA technology grew by leaps and bounds, the notion that guitarists and bass players would make all their own noise stayed entrenched. Hey – they already had the gear, right? Why fix what isn’t busted?

At this same time, the founding fathers of amplified guitar and bass were creating the tones and textures that would define those instruments for decades. They were getting those sounds through gear that had to be big, heavy, and loud to do its job. Especially for the guitar players, who loved (and still do love, for good reason) the thick, satisfying roar of power tubes being driven hard, the acoustical output was in-freaking-sane.

They got away with that volume because it was expected, and also because they were playing to huge crowds. Most of the audience wasn’t in the first few rows, and so the noise wasn’t as deafening.

Now, fast forward to 2013.

The iconic gear that defined the sound of rock and roll instruments is still very much in fashion. Sure, there have been various improvements in materials, construction, cost management, and design, but all of these creatures of the amplifier kingdom are fundamentally the same animals as their counterparts from 1969. They’re big, they’re heavy, and their most rockin’ sounds require stadium volume (or a power soak, if you don’t want stadium volume).

The problem is that stadium volume from amplifiers is no longer required, or even desirable – especially not in small rooms.

The 100 Watt Amp Problem

Let’s talk about some of what’s going on when an all-tube, 100 watt, gorgeous sounding amp is really doing its thing. Let’s make some conservative assumptions to start:

  • The 100 watt rating is the continuous power generated by the amp at a full-tilt, maximum overdrive, supersaturated roar.
  • The cab is a 4×12, wired so that each loudspeaker gets 25 watts.
  • Each loudspeaker has an average sensitivity of 95 dB SPL at 1 watt, measured at 1 meter.

The tone is killer. So is the volume.

Each cone is producing about 109 dB SPL, continuous. The summation of those four cones is 115 dB SPL, continuous, at 1 meter. The average audience member is probably sitting about 22 feet (6.7 meters) away. The venue isn’t totally dead, acoustically, so the average SPL decay is 5 dB per doubling of distance, as opposed to 6. This works out to 13.7 dB of volume decay for the average audience member.

So, for the most part, the audience is hearing about 101 dB SPL, continuous, of just the one guitar. Add another guitar of similar volume, and the continuous level is 104 dB SPL. The bass player fits in with a 99 dB SPL contribution, which takes our total to 105 dB SPL. The drummer is a spirited lad, able to make 100 dB SPL himself. Now we’re at 106 dB SPL. The vocals probably have to be at a minimum continuous level of 102 dB in order to be distinguishable, so that takes us to a grand total of…

Just under 108 dB SPL, continuous, for the average audience member, and that’s not including monitor wash.

For most people, that’s pretty dang loud. In a bar, that kind of level is hard to deal with when placing or taking orders (assuming that the bar is in the “average level” zone – which IS the case in a good number of rooms).

There’s no denying that the tone of the guitar is spectacular, but that spectacular tone is causing an audience discomfort problem, and potentially an economic problem for the venue.

This is bad for you.

Also bad for you is that, to get really good separation, the singer (who’s about 12 feet from the cab) has to be able to produce about 125 dB SPL at their mic capsule. This means that you need a singer with lots of power, stamina, and great pitch control at full volume…or less pitch control, but more raw power in reserve.

On top of that, for the vocalist to feel like they’re really hearing themselves in the monitors, the wedges will need to be making about 115 dB SPL continuous at the singer’s ears. If the singer is really powerful, and the wedges are good, then this should be achievable. If the singer isn’t really powerful, or is having an off day, or if the wedges are a little cheap, getting that kind of level may be a battle. Now, you’ve potentially got gain-before-feedback issues.

The Upshot

That arena-ready amp rig sure does sound good, but:

  • It probably costs a fair amount of money to acquire.
  • It takes up a lot of room.
  • It’s heavy.
  • It has to get really loud before it sounds right.
  • It forces everybody else to keep up.
  • It makes monitors harder to manage.
  • It can drive audience members away.
  • The venue can lose money.
  • It reduces the FOH audio tech’s options for the rest of the band (because the tech’s first priority can be forced towards just keeping up with you).


There’s a fix.

Buy a little amp.

There are plenty of all-tube combos out there that top out at 10 watts. That’s really all that you need. Get those tubes really hot to get the tone you want, and you’ll probably have about 105 – 110 dB SPL at 1 meter.

And you’ll be able to do it with a piece of gear that’s easy to carry.

And you’ll be able to do it with a piece of gear that you can fit anywhere.

And you’ll be able to do it without making your vocalist work themselves to death.

And you’ll be able to do it without forcing everybody else to keep up with you, whether in terms of volume or equipment purposes.

And you’ll be able to do it without flattening the audience.

And you’ll be able to do it while the bar still makes money.

And you’ll be able to do it while allowing the audio tech to make meaningful choices to get you the best sound possible.

And, because PA technology has come a very long way, that one amp will still work for you when you’re playing stadiums. The crew will just stick a mic in front of it, and turn that 10 watt amp into a 10,000+ watt amp with great coverage and smooth frequency response across the entire audience.

I can certainly understand that you might want a big rig because of the way it looks, or because there’s something very specific about the sound that can’t be perfectly replicated by other means. I do get that.

But big amps just aren’t necessary anymore, and they can be more trouble than they’re worth.

Two Speed Limits

The amount of level that an audio rig can deliver is often less than the theoretical maximum.

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.

I don’t think that very many people have trouble understanding the concept of “not enough PA.” For most folks, it just stands to reason that an FOH (Front Of House) or monitor system has a finite amount of level available.

Notice that I said, “most folks.” There are some people out there who do seem to believe that all PA systems are capable of infinite loudness. These people frighten me. They also wreck lots of gear when they’re allowed to drive an audio rig. Audio humans of this particular tribe are genetically related to the tribe of audio humans who must use every ounce of available system power, regardless of whether it’s a good idea or not. “It just doesn’t sound right until I see the clip lights, you know?” *Sigh.*


The idea that audio gear has finite performance limits is pretty intuitive, especially when we’re talking about equipment involved in output transduction (converting electricity to sound pressure waves). What’s not so intuitive is that a system’s maximum practical output is almost always lower than its true maximum.

The True Maximum

The absolute, full-tilt boogie, no-holds-barred maximum SPL that an audio system can produce is most heavily determined by the following:

  • How many drivers you’ve got. This is often simplified into “how many boxes didja bring?” because, most of the time, audio humans deploy drivers that have been conveniently bolted into different kinds of enclosures.
  • How much power you can put into the drivers you have. This can be made more tricky if you don’t actually have enough mains power to drive the amps at full-throttle.
  • The space that the drivers are firing into. Enclosed, highly reflective spaces are “loud” because acoustical energy is reflected back into the room, combining with the “direct” output from the audio system. Outdoors, or in highly absorptive rooms, a much greater proportion of acoustical energy is captured or radiated away from the listeners, never to add to the direct sound.

When you put this all together, you can get yourself some numbers regarding the maximum SPL (sound pressure level) achievable, assuming that no other factor stands in the way. Like I said, though, other factors usually DO stand in the way. One factor is essentially social – that is, how much auditory input people will accept before they perceive the sensation to be unpleasant. This is an important concept, but it’s not really within the scope of this article. However, other major, technically-based limiting factors do apply.


A loudspeaker system might be capable of producing x-amount of SPL, but that doesn’t mean that it can do so in a pleasing fashion. It’s entirely possible that you’ll encounter unacceptably high distortion before you hit the absolute maximum level that a system can produce.

Even though I think most techs unconsciously account for this phenomenon when estimating or empirically determining the maximum output available from a rig, I also think it’s important to mention.

A, *ahem*, “dirty” secret that isn’t necessarily intuitive to folks outside of pro audio is that amplifiers are quite capable of producing more than their “rated” power. Rated power is a number that corresponds to what the amplifier can do, based on a certain amount of distortion that the manufacturer finds “acceptable.” Better manufacturers are less tolerant of distortion. One reputable manufacturer is willing to claim that a certain amplifier can put about 397 watts into two, 8 Ohm loads at less than 0.02% distortion. (You have to do a bit of math, because the actual rated power is 500 watts, and the distortion number is given at 1 dB BELOW rated power, for some reason.)

Distortion that low is pretty hard to hear. According to the calculator at Sengpiel Audio, 0.02% THD is about 73 dB down from the original signal. I rigged up a test with two tones (1 kHz and 2 kHz) in my DAW, and I couldn’t hear the 2 kHz tone (at all) against the 1 kHz signal when the 2k channel was pulled down 73 dB.

Anyway, here’s the deal.

That same amplifier will display greater continuous output if greater distortion is allowed. That 397 watts is a continuous rating, based on an RMS (Root Mean Square) voltage – a bit more than 56 volts. To get that RMS voltage, you need a peak voltage of just a bit less than 79.7 volts. Into an 8 Ohm load, 79.7 volts produces an instantaneous power of almost 800 watts. Take note of that “instantaneous,” though. That 800 watts isn’t applied to the loudspeaker for very long, and so, when everything gets averaged out, the loudspeaker only experiences about 400 watts. (It’s actually more complicated than this, especially with actual music, but this will do for an illustration.)

As you push an amplifier harder and harder, its peak voltage output will remain the same, but the RMS voltage will increase. This is because the amplifier spends more and more time producing output voltages that are closer and closer to the peak voltage.

This power isn’t “free” though. The more you push the amp, the more distortion you get. Some manufacturers will allow for much higher distortion at “rated” power, so as to be able to publish a bigger number on the spec sheet.

Bottom line?

The absolute maximum SPL that you can achieve with a rig under a given set of acoustical conditions may actually require that the system be driven into audible, unpleasant distortion. This distortion isn’t just limited to the amplifiers, either. You could be driving any part of the signal chain too hard, even the loudspeaker drivers themselves. The effective “speed limit” on the rig may be brought down (and brought down a lot) by just how far the system can go before it sounds like a pile of garbage.

Of course, for some folks, this (frighteningly) doesn’t matter very much, leading to the classic pro-audio line of “Well, it sounds like !@#$, but it’s REALLY !@#$%^& loud!”

Gain Before Feedback

Then again, you might never even get to the point of audible distortion. GBF (gain before feedback) issues can lower your effective speed limit even more. The underlying reason for this isn’t necessarily obvious: A given amount of gain only guarantees a repeatable output level if the input signal’s overall level remains unchanged. If the input signal’s overall level changes significantly, any fixed gain is correlated to, but not absolutely matched with a particular system output situation.

In other words, you may have a huge amount of gain applied to something, but if the input signal is very small, the final output can still be very quiet. Now, add in the fact that live audio is almost always a “partially closed loop,” and *WOOOOOOOOoooooossssssquuuuuuEEEEEAALLLL!*


“Feedback” is when the output of a system returns to an input of the system. This can be used for some very cool things, but it can also cause serious problems. In the partially-closed-loop situation of live audio, some fraction of the output of the rig finds its way into a microphone or instrument pickup, and is then re-added to the system output. The parts of the signal that are in phase sum constructively, causing the system output to rise, which means that more signal finds its way back into the system, which means that the system output rises still further at those frequencies, which –


You get the point.

GBF is just a shorthand for “how much can we turn this thing up before it starts to ring.” With some sources, it simply isn’t possible to stay out of feedback while simultaneously applying sufficient gain to drive a rig at full-tilt. There just isn’t enough input.

Where you see this “in the wild” is with the quiet singer (or the singer who’s moderately loud but wants to stand 5 feet from the mic), and also with the quiet player of the [acoustic instrument that you can’t get a good mic placement on]. The same gain structure with a louder source would drive the system all the way to the limiters, while also clipping the console’s input circuitry. However, for these folks, you’re barely making 90 dBC. Maybe not even that.

And yes, you’ve gotten out the EQ and notched the major problem areas. The issue is that any more EQ will either completely wreck the sound of the source, or just reduce the overall gain by the same amount you tried to add.

When you arrive at this kind of situation, you’re at the effective speed limit. You can’t get any more output, because you can’t add any more gain. Turning things up will just result in feedback, terrible sound, or no net gain at all.

As with a lot of things in life, just having the basic capacity to do something doesn’t mean it can actually be achieved. The circumstances have to be right.