Tag Archives: Backline

Compact Can Be Accommodated

When the PA is big and heavy, other things can be small and light.

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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Related: A mouse can fit in a mouse-sized room, a dog-sized room, and an elephant sized room. An elephant can only fit in an elephant-sized room.

Meditate upon this carefully.

There’s also this bit about elephants and garden hoses.

Is The Problem Voltage, Or Voltage Transfer, Or…?

If you’re going to fix a problem, you have to know what the problem actually is.

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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If you’re going to troubleshoot (and if you’re in the business of show production, troubleshooting is inevitable), there are two basic rules:

1) You have to know what the device is supposed to do.

2) You have to know how the device does what it’s supposed to do.

There are many layers of doing 1 and 2 effectively. The deeper you go, the more problem solving you can do. Gaining the knowledge required to peel back more and more layers is a long process. Decades long. I’ve had my hands in pro-audio since I was a teenager, and with about 20 years under my belt I’m finally starting to feel like I get what’s going on. In part, that’s because I’m getting more and more acquainted with the oceans of material I still don’t know. When you start to realize just how deep the rabbit hole is, you’ve been falling down that hole for a good while.

The above is a basic, foundational statement for this article, which is a follow-on to the opening “case study” from my previous post. After having a potential issue discussed with me, I ended up finding an alternate route to a solution. I took the different path because I had a suspicion that the problem wasn’t the voltage level of a pickup’s output. I figured that the real bugbear was that the voltage from the pickup was being transferred poorly, and also that the pickup’s bottom end was being lost. I considered this assumption as possible because I have a notion (not a truly detailed one, but a notion nonetheless) about how high-impedance pickups work. That is, I know that they can be reasonably modeled as a voltage source in series with a capacitance. This all comes together to form a device with a rather high output impedance in pro-audio terms. The issue with high-impedance outputs is that voltage transfer becomes non-trivial, and the issue with capacitors in series with voltage sources is that they create a high-pass filter.

Modeling Voltage Transfer With DC

For audio folks, what we’re interested in is voltage transfer. Even when amplifiers and loudspeakers are involved, and we become interested in power transfer, we achieve power transfer by way of voltage transfer. In many ways, effective voltage transfer is invisible to audio humans. It just sort of happens for us, because a lot of our gear is built to play nicely with a lot of other gear. At times, though, we’ll encounter gear that was NOT actually built to interface nicely with our existing equipment, and that can throw us for a loop. In the case of a high-impedance pickup interfaced with pro-audio inputs, we can get into a situation where we’re PILING on the gain, only to end up with a relatively weak signal. If we don’t know how the device does what it’s supposed to do, then we can start to assume that the voltage from the pickup is too low.

But that’s not the case. Piezo pickups – probably THE example of a high-output impedance device – make plenty of voltage. When mated to, say, a basic DI box, the problem is that the voltage doesn’t transfer. The input impedance of the mic pre is too low.

Before I go any further with this, I need to say something:

IMPORTANT – Audio circuits are NOT direct current. They are alternating current. Modeling an audio circuit via a DC example is not an entirely accurate picture of what’s happening. DC examples are simple to read and easy to “construct,” but they neither show the entire picture nor all the details of what’s happening.

With that in mind…

At a very basic level, the underlying issue with voltage transfer is that voltage drops when it travels across resistors. If we mentally model an audio circuit as a voltage source across one resistor (output impedance), and then have the remaining voltage travel across an additional load resistor (input impedance), we start to get a basic idea of how things can play out.

In our simplified, DC, everything-in-series circuit, the voltages across each resistor add up to the total voltage in the circuit. As such, the proportionality between the resistors representing output and input impedance matters a lot. If the output impedance is high in relation to the input impedance, a good deal of voltage will drop before ever getting a chance to drive the input. In the reverse case, only a small amount of the total voltage drops across the output impedance, allowing a healthy voltage transfer into the next part of the audio chain.

If I take a quick jaunt over to PartSim, I can build a quick ‘n dirty example circuit. This one represents one of my EV ND767a mics plugged into one of the preamps they usually “see,” which are on an M-audio Profire 2626. At a continuous level of 94 dB SPL (1 Pascal), an ND767a is rated for 3.1mV of RMS voltage output. That output can be modeled as being in series with a 300 ohm resistor. The mic-pre of the Profire can be modeled as a 3.7 kilohm resistor.


In this example, 0.23mV drops across the output impedance of the microphone. If you do the math to figure out the decibel loss, you find that about 0.67 dB was lost before the signal hit the mic pre. Even with this being a DC example, that number tracks very well with the output of the bridging calculator at Sengpiel Audio.

The above is an example of equipment that’s designed to interface nicely. What happens when a piezo pickup gets plugged into a basic DI box? That’s probably something like a 1 megohm output impedance being mated to a 50 kilohm input. The piezo can develop plenty of electrical potential. One volt RMS is +2.2 dBu, or definitely within the “line level” range. The voltage isn’t a problem at all, but the transfer of that voltage is a big deal.


Immediately, 26 dB of voltage is dropped. If the DI box steps the voltage down even further (as is apt to happen), then the signal arriving at the console pre might be 46 dB down from the original voltage supplied by the pickup. The voltage arriving at the preamp is no higher than what you would get from a “hot output” dynamic mic in front of a not-too-loud source.

But Why Does It Sound So Bad?

Now then.

If the only real downside of our “not enough input impedance” situation was voltage loss, it wouldn’t be so bad. We’d have to run our preamps a little hot, but that’s hardly a dealbreaker.

The real awfulness comes about when the AC circuit issues enter into play. As I mentioned earlier, a piezo pickup in an audio circuit naturally tends to create a high-pass filter on its output. The high-pass filter becomes less audible as the load (input) impedance goes up. The problem, then, is that a too-small load impedance causes a very marked loss of low-frequency information. The pickup sounds “clanky” or “nasal,” because all of its really usable output becomes restricted to the high-frequency part of the audio passband.

Here’s a simplified model of a piezo pickup connected to a 50 kilohm DI box. I haven’t tried to fully represent the output impedance of the pickup, so the voltage numbers won’t be right. I used a 650 pF capacitor to represent the pickup, because the simulation of the circuit with that capacitance seems to basically represent what I’ve observed in the field.



At 1 kHz, the signal is about 13 dB down from the maximum level. At 200 Hz the signal is down 27 dB. Good luck correcting that with any bog-standard EQ you have handy.

Compare that with what happens when the load impedance is 1 megohm, which is what some of my active DI boxes are rated for:


Yes, there’s still a highpass filter in effect. Even so, it’s rather less terrifying. The filter’s 3 dB down point happens at about 250 Hz, and you’re only about 8 dB down at 100 Hz. That’s hardly perfect, but it’s manageable. (A DI box or preamp with a 10 megohm input impedance basically makes the low frequency loss a non-issue.)

Once more, I need to emphasize that these are simple models. They won’t exactly represent what you run into during the course of setting up an actual show.

But they do show that the voltage generated by a troublesome audio source is not necessarily the root of a given problem. Poor voltage transfer and circuits that mess with frequency response (when presented with a small load impedance) may be what’s really hurting you.

Mysteriously Clean

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

Please Remember:

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

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

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

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

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

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

Let me explain.

Drums, Drums, Everywhere

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


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

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

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

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

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

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

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

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

Directivity Matters

Now then.

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

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

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


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

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

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

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

Vocal Processors, And The Most Dangerous Knob On Them

If you were wondering, the most dangerous knob is the one that controls compression.

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.

Not every noiseperson is a fan of vocal processors.

(Vocal processors, if you didn’t know, are devices that are functionally similar to guitar multi-fx units – with the exception that they expect input to come from a vocal mic, and so include a microphone preamp.)

Vocal processors can be deceivingly powerful devices, and as such, can end up painting an audio-human into a corner that they can’t get out of. The other side of that coin is that they can allow you to intuitively dial up a sound that you like, without you having to translate your intuitive choices into technical language while at a gig.

What I mean by that last bit is this: Let’s say that you like a certain kind of delay effect on your voice. There’s a specific delay time that just seems perfect, a certain number of repeat echoes that feels exactly right, an exact wet/ dry mix that gives you goosebumps, and an effect tonality that works beautifully for you. With your own vocal processor, you can go into rehearsal and fiddle with the knobs for as long as it takes to get exactly that sound. Further, you don’t have to be fully acquainted with what all the settings mean in a scientific sense. You just try a bit more or less of this or that, and eventually…you arrive. If you then save that sound, and take that vocal processor to a gig, that very exact sound that you love comes with you.

Which is great, because otherwise you have to either go without FX, or (if you’re non-technical) maybe struggle a bit with the sound person. The following are some conversations that you might have.

You: Could I have both reverb and delay on my vocal?

FOH (Front Of House) Engineer: Ummm…we only have reverb.

You: Oh.

You: Gimme a TON of delay in the monitors.

Audio Human: Oh, sorry, my FX returns can only be sent to the main mix.

You: Aw, man…

You: Could I have a touch more mid in my voice?

[Your concept of “a touch more mid” might be +6 dB at 2000 Hz, with a 2-octave-wide filter. The sound-wrangler’s concept of “a touch more mid” might be +3 dB at 750 Hz, with a one-octave-wide filter. Further, you might not be able to put a number on what frequency you want, especially if what I just said sounds like gobbledygook. Heck, the audio human might not even be able to connect a precise number with what they’re doing.]

Sound Wrangler: How’s that?

You: That’s not quite right. Um…

[This one’s directly in line with my original example.]

You: Could I get some delay on my voice?

Audio Human: Sure!

[The audio human dials up their favorite vocal-delay sound.]

You: Actually, it’s more of a slap-delay.

[Your concept of slap-delay might be 50 ms of delay time. The audio-human’s concept of slap-delay might be 75 ms.]

Audio Human: How’s that?

You: That’s…better. It’s not quite it, though. Maybe if there was one less repeat?

[The audio-human’s delay processor doesn’t work in “repeats.” It works in the dB level of the signal that’s fed back into the processor. The audio-human takes a guess, and ends up with what sounds like half a repeat less.]

Audio Human: Is that better?

You: Yeah, but it’s still not quite there. Um…

Having your own vocal processor can spare you from all this. It also spares the engineer from having to manage when the FX should be “in” or bypassed. (This often isn’t a huge issue, but it can become one if you’re really specific about what you want to happen where.) There are real advantages to being self-contained.

There are negative sides, though, as I alluded to earlier. Having lots of power at your disposal feels good, but if you’re not well-acquainted with what that power is actually doing, you can easily sabotage yourself. And your band. And the engineer who’s trying to help you.

EQ Is A Pet Dog

The reason that I say that “EQ is a pet dog” is twofold.

1) EQ is often your friend. Most of the time, it’s fun to play with, and it “likes” to help you out.

2) In certain situations, an EQ setting that was nice and sweet can suddenly turn around and “bite” you. This isn’t because EQ is “a bad dog,” it’s because certain equalization tweaks in certain situations just don’t work acoustically.

What I’ve encountered on more than one occasion are vocal-unit EQ settings that are meant to either sound good in low-volume or studio contexts. I’ve also encountered vocal-unit EQ that seems to have been meant to correct a problem with the rehearsal PA…which then CAUSES a problem in a venue PA that doesn’t need that correction.

To be more specific, I’ve been in various situations where folks had a whole busload of top-end added to their vocal sound. High-frequency boosts often sound good on “bedroom” or “headphone” vocals. Things get nice and crisp. “Breathy.” Even “airy,” if I dare to say so. In a rehearsal situation, this can still work. The rehearsal PA might not be able to get loud enough for the singer to really hear themselves when everybody’s playing, especially if feedback can’t be easily corrected. However, the singer hears that nice, crisp vocal while everybody’s NOT playing, and remembers that sound even they get swamped.


The problem with having overly hyped high-end in a live vocal (especially with a louder band in a small room) is really multiple problems. First, it tends to focus your feedback issues into the often finicky and unpredictable zone of high-frequency material. If there’s a place where both positionally dependent and positionally independent frequency response for mics, monitors, and FOH speakers is likely to get “weird” and “peaky,” the high-frequency zone is that place. (What I mean by “positionally dependent” is that high-frequency response is pretty easy to focus into a defined area…and what THAT means is that you can be in a physical position where you have no HF feedback problems, and then move a couple of steps and make a quarter turn and SQUEEEEAAALLL!)

The second bugbear associated with cranked high-end is that, when the vocals are no longer isolated, the rest of the band can bleed into the vocal mic LIKE MAD. That HF boost that sounds so nice on vocals by themselves is now a cymbal and guitar-hash louder-ization device. If we get into a high-gain situation (which can happen even with relatively quiet bands), what we then end up doing is making the band sound even louder when compared to your voice. If the band started out a bit loud, we may just have gotten to the audience’s tipping point – especially since high-frequency information at “rock” volume can be downright painful. Further, we’re now spending electrical and acoustical headroom on what we don’t want (more of the band’s top end), instead of what we do want (your vocal’s critical range).

Now, I’m not saying that you can’t touch the EQ in your vocal processor, or that you shouldn’t use your favorite manufacturer preset. What I am saying, though, is that dramatic vocal-processor EQ can really wreck your day at the actual show. You might want to find a way to quickly get the EQ bypassed or “flattened,” if you can.

“Compression” Is The Most Dangerous Knob On That Thing

Now, why would I say that, especially after all my ranting about EQ?

Well, it’s like this.

An experienced audio tech with flexible EQ tools can probably “undo” enough of an unhelpful in-the-box equalization solution, given a bit of time. Compression, on the other hand, really can’t be fully “undone” in a practical sense in most situations. (Yes – there is a process called “companding” which involves compression and complementary expansion, but to make it work you have to have detailed knowledge of the compression parameters.) Like EQ, compression can contribute to feedback problems, but it does so in a “full bandwidth” sense that is also much more weird and hard to tame. It can also cause the “we’re making the band louder via the vocal mic” problem, but in a much more pronounced way. It can prevent the vocalist from actually getting loud enough to separate from the rest of the band – and it can even cause a vocalist to injure themselves.

Let’s pick all that apart by talking about what a compressor does.

A compressor’s purpose is to be an automatic fader that can react at least as quickly (if not a lot more quickly) as a human, and that can react just as consistently (if not a lot more consistently) as a human. When a signal exceeds a certain set-point, called the threshold, the automatic fader pulls the signal down based on the “ratio” parameter. When the signal falls back towards the threshold, the fader begins to return to its original gain setting. “Attack” is the speed that the fader reduces gain, and “release” is the speed that the fader returns to its original gain.

Now, how can an automatic fader cause problems?

If the compressor threshold is set too low, and the ratio is too high, the vocalist is effectively pulled WAY down whenever they try to deliver any real power. If I were to set a vocalist so that they were comfortably audible when the band was silent, but then pulled that same vocalist down 10 dB when the band was actually playing, the likely result with quite a few singers would be drowned vocals. This is effectively what happens with an over-aggressive compressor. The practical way for the tech to “fight back” is to add, say, 10 dB (or whatever) of gain on their end – which is fine, except that most small-venue live-sound contexts can’t really tolerate that kind of compensating gain boost. In my experience, small room sound tends to be run pretty close to the feedback point, say, 3-6 dB away from the “Zone of Weird Ringing and Other Annoyances.” When that’s the case, going up 10 dB puts you 4-7 dB INTO the “Zone.”

But the thing is, the experience of that trouble area is extra odd, because your degree of being in it varies. When the singer really goes for it, the processor’s compressor reduces the vocal mic’s gain, and your feedback problem disappears. When they back off a bit, though, the compressor releases, which means the gain goes back up, which means that the strange, phantom rings and feedback chirps come back. It’s not like an uncompressed situaton, where feedback builds at a consistent rate because the overall gain is also consistent. The feedback becomes the worst kind of problem – an intermittent one. Feedback and ringing that quickly comes and goes is the toughest kind to fight.

Beyond just that, there’s also the problem of bleed. If you have to add 10 dB of gain to a vocal-mic to battle against the compressor, then you’ve also added 10 dB of gain to whatever else the mic is hearing when the vocalist isn’t singing. Depending on the situation, this can lead to a very-markedly extra-loud band, with all kinds of unwanted FX applied, and maybe with ear-grating EQ across the whole mess. There’s also the added artistic issue of losing dynamic “swing” between vocal and instrumental passages. That is, the music is just LOUD, all the time, with no breaks. (An audience wears down very quickly under those conditions.) In the circumstance of a singer who’s not very strong when compared to the band, you can get the even more troublesome issue of the vocal’s intelligibility being wrecked by the bleed, even though the vocal is somewhat audible.

Last, there’s the rare-but-present monster of a vocalist hurting themselves. The beauty of a vocal processor is that the singer essentially hears what’s being presented to the audience. The ugliness behind the beauty is that this isn’t always a good thing. Especially in the contexts of rock and metal, vocal monitors are much less about sounding “hi-fi” and polished, and much more about “barking” at a volume and frequency range that has a fighting chance of telling the singer where they are. Even in non-rock situations, a vital part of the singer knowing where they are is knowing how much volume they’re producing when compared to the band. The most foolproof way for this to happen is for the monitors to “track” the vocalists dynamics on a 1:1 basis – if the singer sings 3 dB louder, the monitors get 3 dB louder.

When compression is put across the vocalist immediately after the vocal mic, the monitors suddenly fail to track their volume in a linear fashion. The singer sings with more power, but then the compressor kicks in and holds the monitor sound back. The vocalist, having lost the full volume advantage of their own voice plus the monitors, can feel that they’re too quiet. Thus, they try to sing louder to compensate. If this goes too far, the poor singer just might blow out their voice, and/ or be at risk for long-term health issues. An experienced vocalist with a great band can learn to hear, enjoy, and stop compensating for compression…but a green(er) singer in a pressure situation might not do so well.

(This is also why I advocate against inserting compression on a vocal when your monitor sends are post-insert.)

To be brutally honest, the best setting for a vocal-processor’s compressor is “bypass.” Exceptions can be made, but I think they have to be made on a venue-to-venue, show-to-show basis.

All of this might make it sound like I advocate against the vocal processor. That’s not true. I think they’re great for people in the same way that other powerful tools are great. It’s just that power tools can really hurt you if you’re not careful.

“It Was On Sale” Is A Bad Reason

A great price on something that doesn’t work for you is not a good deal.

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.

The wrong gear at the right price is still the wrong gear.

More Is More…And More Of Everything Connected To That More

A Small Venue Survivalist Saturday Suggestion

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.

The guy with the most toys…

…has the longest load-in.

And load-out, but one thing at a time, okay?

It’s Called “A Band”

A Small Venue Survivalist Saturday Suggestion

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.

This thing that you’re involved in is called “a band.”

It is not called “the sound of your amp’s power tubes saturating while a few other people hang out on stage for no discernible reason.”

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.

The Elephant vs. The Garden Hose

If you make your production fit a small venue, it will fit anywhere.

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.

A while ago, I was a participant on Harmony Central’s live-sound forums. On those forums, a few people emerged as authoritative, experienced professionals who could be counted on as voices of reason. One of those people was W. M. Hellinger. Besides being the proprietor of audiopile.net (one of my favorite places to shop for audio cable), he was a regular dispenser of plain-old, hard-won, good sense. In my recollection, Mr. Hellinger could almost always be counted on to provide some gem of “in-the-trenches” wisdom, often related as an anecdote or amusing metaphor. One of his most memorable was offered in a thread regarding methods for working with a loud drummer:

“…if you’re having problems trying to stuff an elephant down a gardenhose, then either the elephant is too large or the garden hose is too small or maybe elephants were never meant to be stuffed down garden hoses, and it’s time to re-think the project.” Full context available here.

Poetry. Audio-cowboy poetry, but poetry just the same.

…and, like many other great metaphors, there’s a lot of meaning packed into it.

Elephants Are Hard To Compact

On several occasions, I’ve encountered acts with what I’ve come to call “Warped Tour-itis.” I don’t have anything against Warped Tour, or the bands that play on it, but I swear that the groups I’ve had the most struggles with would fit into the event perfectly. These are the bands that are trying to fit an auditorium or shed show into an enclosed place that seats 200 people or fewer.

Their show was built to be the size of an elephant (in one way or another), and when faced with a “garden hose” of a venue, there’s no way to downscale. They just try to get through by force of will.

Now, sure, their physical setup might fit, but what usually doesn’t fit is the volume.

  • The drummer hits as hard as he possibly can. Especially the cymbals. All the time. His snare sounds like a firearm. (I’m not joking. One of these guys once smacked his snare while standing behind me, and I swear that it sounded like he had discharged a pistol.) It’s all about “being intense” and having “great energy” – which would be super fun if it didn’t hurt to be in the room with the guy.
  • The guitar players have all-tube heads, which sport big wattage. Those heads are connected to either a half or full-stack of cabs, and the rigs “just don’t sound right” if dialed back to anything less than “crushing.” Of course, if they could dial their amps back, they’d just get run over by the drummer. Even so, they want a lot of their rig in the their monitor wedge. And a lot of the other guitar player. And a lot of kick and snare, because they can’t hear the drums anymore. Plus some bass.
  • The bass player has at least one 8×10, powered by a massive head. It’s Ampeg, of course. The head is vintage, vintage being a synonym for “runs hot and weighs as much as the trailer it rides in.” The amount of energy produced by the bass stack is formidable. The bottom octave is felt as much as heard. Whatever subs are available to the PA, they’re overmatched by 6-10 dB.
  • The vocalist has to do the “scream” thing. There is literally no other option, except for when the guitarists have switched to their clean channel. At any other time, vocal-chord threatening volume is required.

Anyway, you get the point.

The band would sound great if they were in an open-air venue, and the average listener was a minimum of 50 feet from the barricade. In a small space, though, the results are uncomfortable. Or downright deafening.

…and the thing is, the “elephant” can’t be compacted.

The drummer’s kit is built specifically to be a certain volume. He can’t switch for a quieter snare, for instance, because he only has the “holy grail” snare that he poured all his money into. His muscle-memory is built around playing at full tilt, with sticks of a certain weight. It’s almost impossible for him to “turn down.”

The guitar rigs, in the same way, are built to get a certain tone at a certain level. In all likelihood, the guitarists invested all their setup money into those stacks. They have no alternative but to use them, and even with master volume controls onboard, they have to keep up with the (essentially fixed volume) drummer.

It’s the same for the bass player, because he has to keep up with the guitars, and it’s not as though the vocalist can scream at a “front-parlor appropriate” volume.

The elephant simply can not be scaled down to fit the garden hose – not at a moment’s notice, anyway.

The faulty logic in play is “if we create a show that works at large scale, then we’re ready for anything.” This seems reasonable, but it’s actually incorrect. It’s a forgivable mistake, because I’m fairly sure that all of us in live-music have made it. We assume that the band is the vehicle that carries the show, and a huge vehicle can carry any size of show. The truth is that the show (or, more correctly, the show’s context) is the vehicle for the band, and a band that’s too “large” will overwhelm the vehicle.

So – what to do?

Elephants Are Remarkably Easy To Expand

If the situation really is that you have to fit into a variety of garden hoses, then the solution is simple:

Make the elephant small enough to fit through the smallest hose you’re going to encounter.

If your show can run comfortably in a small club, then a competent crew can scale that show up to auditorium or shed-gig size when it comes time. When you get invited to that big show, the show itself will have the resources necessary to make the act big enough to fll a much larger garden hose.

If you’ve invested your time and money into a drumkit and play style that works nicely in a small club, you can be mic’ed up and reinforced for pretty much any number of people. When it’s time to play in a tiny room again, all you have to do is what you’ve always done.

If you’ve invested in a guitar or bass rig that sounds great at small-venue volume, then you’ll sound just as great when the amp gets sent through a PA sized appropriately for the show. (I’m not kidding. One of the biggest, most raging-awesome “PanterrrrRRRRAAAAAA!” guitar tones I’ve ever heard was the result of micing a Roland cube. I’ve had lunch boxes bigger than that thing.)

If your show is exciting, and yet manageable in a space the size of a postage stamp, then there won’t be any insurmountable issues to be found in making it happen on a huge stage. Sure, you might not take advantage of the whole area all at once, but that’s not what actually makes shows great.

The bottom line is that show production – audio, lighting, staging, logistics, whatever – is primarily an additive activity. Making things larger than life is pretty much what all the technology is built around, because that’s how the laws of physics work. Subtractive techniques are few in number and difficult to implement.

Small elephants fit down small garden hoses, and when you’re just starting out, the small hoses are what you’ll need to fit your show into. Small doesn’t mean “dinky” or “boring.” It just means compact.

So, build the most amazing, travel-sized elephant that you possibly can.

Why Buy An Active DI

An active DI box can cost a bit more, but they have big advantages.

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.

The beat-up device up there with a missing knob is one of my well-loved passive DI boxes. It’s sounded plenty decent on a number of sources, solved at least its own share of ground-loop issues, and has never had any problems (beyond losing its attenuator knob).

Passive DI boxes are very handy creatures. They solve connectivity problems with almost no fuss at all, and the well-designed models are highly resistant to both stupidity and malice. My guess is that, about 80% of the time, they’re a perfectly decent choice.

The thing is, though, that active DI boxes let you cover the full 100% at all times. They’re also cheap enough now that there’s really no reason not to go active (if you’re starting from scratch).

The “cheap enough” bit is pretty self-explanatory. Head on over to your favorite music-gear retailer – Sweetwater, PSSL, Zzounds, whoever – and find their direct box category. Sort by ascending price, and you’re almost sure to find active units before you leave the $30 price point. (Some of the really cheap units are junky, but to be fair, I own two Behringer DI800 units that have never let me down…and at $120 a pop, their per-channel cost is $15.)

What isn’t so self-explanatory is why passive units don’t quite cover 100% of the direct-input situations you’ll encounter. There’s a bit of science involved.

A Few 10s of kOhms Is Usually Enough

Modern audio is all about voltage transfer. Voltage transfer is all about connecting an output device to an input device with an impedance (opposition to current flow) that is high when compared to the output circuit.

Okay, that sounds like gobbledygook. An analogy would be helpful.

Think of a bunch of cars on the freeway. Traffic is flowing nicely. Everybody’s just flying along without a care in the world. This is low impedance. There’s very little opposition to traffic flow.

Now, we construct an exit to the freeway. The exit leads to a one-lane road. The one-lane road, in comparison to the freeway, is a high-impedance device. Fewer cars can flow down that one lane road, and as a result, the freeway has no trouble keeping the little road supplied with cars.

This condition, when applied to electrical connections, is called “bridging impedance.” An output device with low impedance is like a freeway, and an input device with a comparatively high (10x or more) impedance is like a one-lane road. For audio types, we’re not concerned with preserving the amount of electrical flow, so much as we’re concerned with preserving electrical force (voltage). Bridging impedance lets us do that.

Most passive DI boxes have an input impedance that’s in the range of several tens of thousands of Ohms. Some can even be in the 100,000 Ohm range. Connect a device with an output impedance of a few thousand Ohms or less, and – no problem! A lot of devices are perfectly suited to interacting with a passive DI, because a lot of the gear and instruments that get connected are active units. Keyboard outputs are low-impedance creatures. Guitar-processors have low-impedance outputs.

Heck, a lot of acoustic-electric guitar outputs are low impedance. The actual pickup might be anything under the sun, but quite often you’ll find some sort of preamp sitting between the pickup and the output jack.

In a lot of cases, you can even get away with connecting a bass or electric guitar with passive pickups to a passive DI. It’s not theoretically ideal, but it usually sounds fine.

This covers the “80% of the time” thing. The 20% comes in when you encounter an instrument with a very high impedance pickup, and no preamp. Plug one of those into a passive DI, and…yuck.

Easy As Pie-zo. (Yeah, That Was A Cheesy Pun…)

The ur-example of the high-impedance pickup is the piezo. Piezo pickups are neat because they’re small, put in direct contact with the instrument (which makes them resistant to external noises, insofar as the instrument resists those noises), affordable, and simple.

The problem with piezos is that they are passive devices with a very high output impedance – so high that getting into impedance bridging territory requires millions of Ohms or more.

So, you plug one of the little darlings into a passive DI, and what happens?

First, you probably get a weak signal out of the pickup. Poor impedance bridging means poor voltage transfer, and voltage transfer is how we ensure good signals in the world of pro-audio.

Second, the instrument probably sounds terrible.


A piezo pickup (when connected to another audio device and viewed as a set of electrical building blocks) is a capacitor, inductor, and load resistor in series, with a capacitor connected in parallel before the load resistor.

What all of that means is that passive EQ is happening – the capacitor, inductor, and load form a classic resonant circuit. The capacitor and inductor in series allow a range of frequencies through, and the parallel capacitance acts as an additional low-pass filter. (Whether or not this low-pass is significant after the capacitor-inductor bandpass is a whole other issue.)

The issue with passive filter circuits is that everything has an effect on everything else. If the load impedance is adequately high, then we get a nicely damped, wideband filter that sounds natural. If the load impedance is too low, however, the filter gets narrow and odd sounding. This effect can become so pronounced that string instruments start to sound like horns(!)

The obvious fix, then, is to connect the piezo pickup to a very-high impedance device. An easy way to do this is to use an active DI box.

The Buffer Zone

Active DI boxes solve the piezo impedance problem because they can employ buffer amplifiers. The great thing about a buffer amplifier is that its input impedance is very, very high (millions or even billions of Ohms). It also does this in a very small package. You could probably construct a passive DI box with an input impedance in the millions of Ohms, but the size and weight of the thing (not to mention the cost) would be really off-putting.

The downside of using a buffer amplifier is that it requires a power supply. This means batteries, or engaging phantom power from the console. In practical reality, though, this downside is almost negligible. Almost any modern console that’s capable of mixing a full band will have phantom available, and a battery in a DI box will probably last for tens (if not a hundred or so) hours.

So – all of this is just a very long way of saying, “Buy active DI boxes.” They’re pretty much guaranteed to work with any kind of instrument output you encounter, and they can be powered by any half-decent console or mic pre. They remove any need for guesswork, and they can even have nifty extras like signal boosters and guitar cab emulations.

Passive direct boxes are the right choice most of the time, but a reliable, full-featured, active DI is the right choice all the time.

No contest.