Category Archives: Gear for Sound and Lighting

Reviews and opinions regarding audio and lighting equipment.

No, Analog Isn’t Better

Analog gear does look cool, though.

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.

Although the fight isn’t nearly so pitched as it once was, some folks might still ask: “Is analog better than digital?”

Analog audio gear does indeed have one major advantage over its number-crunching counterparts. Especially with the right lighting, it often looks a lot cooler on Instagram. Other than that, I’ll take digital over analog any day of the week, and twice on Sunday.

Everyone’s got their own opinion, of course, and I can respect that. I believe that I can back mine up pretty convincingly.

“Back in the day,” you could make a case that analog sounded better. I maintain that this was because both analog and digital grunged up signals to about the same degree, but that digital grunge is generally perceived as being less pleasing. We’re in the 21st Century now, though, and those problems were fixed a good while back. Today’s digital is clear, hyper-accurate, and pristine, even with all manner of gain-changes piled on and low-level signals being passed. Along with that, digital gear is compact, lightweight, flexible, cheap, and feature rich.

Analog, on the other hand, is large, heavy, inflexible, expensive, and feature-limited. It also does not sound “better.”

What do I mean?

Let’s take the example of a modern, digital console, like an X32 Core. Such a console is the ultimate expression of digital’s strengths:

First of all, the setup is tiny. With six rack-spaces handy, you can have 32 X 16 I/O, plus a separate console for FOH and monitor world. Of course, the system has no control surface, so you’ll need a laptop or tablet to act as a “steering wheel.” Even so, the whole shebang could fit in the trunk of a small car. A similar analog setup would necessitate a good-sized SUV, truck, or van for transport.

This also factors into the lightweight aspect. I don’t know exactly how much the above system weighs, but I know it’s a LOT less than two, 32 input analog boards. Even with no other accoutrements, the old-school solution will put you into the 80-pound range at a minimum. Add in a traditional multicore and stagebox splitters, and…well…it’s a lot to carry.

The flexibility argument comes next. Although everything has a design limit, gear that runs on code can have updates applied easily. As long as any new functionality falls within what the hardware and basic software platform can manage, that new functionality can be added – through a simple software update – for as long as the manufacturer cares to work on the system. Front-end control is just as malleable, if not more. If it turns out that the software portion of the interface could do things better, an update gets written and that’s that. Equipment that operates on physical circuits either has no path for similar changes, or if it does, accomplishing the changes is a task that’s profoundly difficult in comparison.

Cost and feature-set dovetail into one another. At the very bare minimum, you can purchase the mixers for a dual-console analog system for about $2800. That’s not too bad in the grand scheme of things, until you realize that a similar investment in the digital world can also get you the stagebox and snake. Also, the digital system will have tons of processing muscle that the analog setup won’t be able to touch. Twelve monitor mixes, fully-configurable channel-per-channel dynamics, four-band parametric EQ, a sweepable filter, EQ and dynamics on every output, plus eight additional processing units? Good luck finding that in an integrated analog package. Such a thing doesn’t even exist as far as I know, and anything even remotely comparable won’t be found for less than tens of thousands of dollars.

So, what about my last point? That analog doesn’t actually sound better?

It doesn’t. No, really. It may sound different. You may like that it sounds different. I can’t argue with personal taste. The reality, though, is that the different sound (especially “warmth” or “fatness” or “depth”) is the product of the gear not passing a clean signal. Maybe the circuitry imparts a nice, low-frequency bump somewhere. Maybe it rolls off in the highs. Maybe there’s just a touch of even-harmonic distortion that creeps in at your preferred gain structure. That’s nifty, but in any objective sense it’s either a circuit that’s inflexibly pre-equalized or is forgiving when being run hard. That may be what some people want, but it’s not what I want, and I’m not going to label it as “better” when a pleasing result is precipitated by a design limitation. (Or only appears when the gain is set just-so.)

Analog isn’t dead, and it isn’t going to die. Our digital systems require well-designed analog stages on the input and output sides to function in real life. At the same time, there are good reasons to make as much of the signal chain digital as is possible. Digital sounds great, and holds too many practical advantages for it to lose out in an objective comparison.


Hitting The Far Seats

A few solutions to the “even coverage” problem, as it relates to distance.

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 article, like the one before it, isn’t really “small venue” in nature. However, I think it’s good to spend time on audio concepts which small-venue folk might still run across. I’m certainly not “big-time,” but I still do the occasional show that involves more people and space. I (like you) really don’t need to get engaged with a detailed discussion regarding an enormous system that I probably won’t ever get my hands on, but the fundamentals of covering the people sitting in the back are still valuable tools.

This article is also very much a follow up to the piece linked above. Via that lens, you can view it as a discussion of what the viable options are for solving the difficulties I ran into.

So…

The way that you get “throw” to the farthest audience members is dependent upon the overall PA deployment strategy you’re using. Deployment strategies are dependent upon the gear in question being appropriate for that strategy, of course; You can’t choose to deploy a bunch of point-source boxes as a line-array and have it work out very well. (Some have tried. Some have thought it was okay. I don’t feel comfortable recommending it.)

Option 1: Single Arrival, “Point Source” Flavor

You can build a tall stack or hang an array with built-in, non-changeable angles, but both cases use the same idea: Any given audience member should really only hear one box (per side) at a time. Getting the kind of directivity necessary for that to be strictly true is quite a challenge at lower frequencies, so the ideal tends to not be reached. Nevertheless, this method remains viable.

I’ve termed this deployment flavor as “single arrival” because all sound essentially originates at the same distance from any given audience member. In other words, all the PA loudspeakers for each “side” are clustered as closely as is practical. The boxes meant to be heard up close are run at a significantly lower level than the boxes meant to cover the far-field. A person standing 50 feet from the stage might be hearing a loudspeaker making 120 dB SPL at 3 feet, whereas the patrons sitting 150 feet away would be hearing a different box – possibly stacked atop the first speaker – making 130 dB SPL at 3 feet. As such, the close-range listener is getting about 96 dB SPL, and the far-field audience member also hears a show at roughly 96 dB SPL.

This solution is relatively simple in some respects, though it requires the capability of “zone” tuning, as well as loudspeakers capable of high-output and high directivity. (You don’t want the up-close audience to get cooked by the loudspeaker that’s making a ton of noise for the long-distance people.)

Option 2: Single Arrival, Line-Array Flavor

As in the point source flavor, you have one array deployed “per side,” with each individual box as close to the other boxes as is achievable. The difference is that an honest-to-goodness line-array is meant to work by the audible combination of multiple loudspeakers. At very close distances, it may be possible to only truly hear a small part of the line, and this does help in keeping the nearby listeners from having their faces ripped off. However, the overall idea is to create a radiation pattern that resembles a section of a cylinder. (Perfect achievement of such a pattern isn’t really feasible.) This is in contrast to point-source systems, where the pattern tends towards a section of a sphere.

As is the case in many areas of life, everything comes down to surface area. A sphere’s surface area is 4*pi*radius^2, whereas the lateral surface area of a cylinder is 2*pi*radius*height. The perceived intensity of sound is the audible radiation spread across the surface area of the radiation geometry. More surface area means less intensity.

To keep the calculations manageable, I’ll have to simplify from sections of shapes to entire shapes. Even so, some comparisons can be made: At a distance of 150 feet, the sound power radiating in a spherical pattern is spread over a surface area of 282,743 square feet. For a 10-foot high cylinder, the surface area is 9424 square feet.

For the sphere, 4 watts of sound power (NOT electrical power!) means that a listener at the 150 foot radius gets a show that’s about 71 dB. For the cylinder, the listener at 100 feet should be getting about 86 dB. At the close-range distance of 50 feet, the cylindrical radiation pattern results in a sound level of 91 dB, whereas a spherical pattern gets 81 dB.

Putting aside for the moment that I’m assuming ideal and mathematically easy conditions, the line-array has a clear advantage in terms of consistency (level difference in the near and far fields) without a lot of work at tuning individual boxes. At the same time, it might not be quite as easily customizable as some point-source configurations, and a real line-source capable of rock-n-roll volume involves a good number of relatively expensive elements. Plus, a real line has to be flown, and with generous trim height as well.

Option 3: Multiple Arrival, Any Flavor

This is otherwise known as “delays.” At some convenient point away from the main PA system, a supplementary PA is set. The signal to that supplementary PA is made to be late, such that the far system aligns pleasingly with the sound from the main system. The hope is that most people will overwhelmingly hear one system over the other.

The point with this solution is to run everything more quietly and more evenly by making sure that no audience member is truly in the deep distance. If each PA only has to cover a distance of 75 feet, then an SPL of 90 dB at that distance requires 118 dB at 3 feet.

The upside to this approach is that the systems don’t have to individually be as powerful, nor do they strictly need to have high-directivity (although it’s quite helpful in keeping the two PA systems separate for the listeners behind the delays). The downside is that it requires more space and more rigging – whether actual rigging or just loudspeakers raised on poles, stacks, or platforms. Additionally, you have to deal with more signal and/ or power runs, possibly in difficult or high-traffic areas. It also requires careful tuning of the delay time to work properly, and even then, being behind or to the side of the delays causes the solution to be invalid. In such a condition where both systems are quite audible, the coherence of the reproduced audio suffers tremendously.


If I end up trying the Gallivan show again, I think I’ll go with delays. I don’t have the logistical resources to handle big, high-output point-source boxes or a real array. I can, on the other hand, find a way to boxes up on sticks with delay applied. I can’t say that I’m happy about the potential coherence issues, but everything in audio is a compromise in some way.


How Could 10 Watts Be Too Loud?

We think audiences want volume, but I’m not sure that’s really true.

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.

I’m not just hammering on players here. The context for this is very much “pro-sound.”

I used to have this regular gig that I loved dearly. Fats Grill is now a hole in the ground, but just a couple of years ago we had live-music every weekend. The PA in the downstairs venue was anything but huge, and yet it was very, very adequate for the space. The mid-highs were mated to an amplifier capable of putting 1000-watt peaks into each box. That works out to a theoretical 127 dB SPL peak for each enclosure – if only at close range (1 meter).

If you were in the middle of the room, you were about 4 meters (or 13-ish feet) away. We’ll say that makes for a practical peak of 115 dB SPL per mid-high, although the room being tightly enclosed would make the real number around 118. Put the two boxes together, and you had a system that could deliver a 121 dB peak in the midrange, plus whatever the subs could do.

Now then.

In pro-audio terms, a 121 dB peak isn’t considered “really loud.” It’s especially not considered loud when you realize that the continuous level, or what humans hear readily, was about 10 dB below that.

But here’s the thing: My experience suggests to me strongly that most folks don’t really want their live-music as loud as “music people” might think. Even for those that love their Rock and/ or Roll, 111 dB continuous can be considered bombardment. This is especially true for the 100 Hz – 15 kHz range. (Subwoofer material is far more easily tolerated, generally speaking.)

At Fats, I very regularly had the system limited so that the top boxes hit a brick wall at their amplifier’s -10 dB point. That’s a peak output of 111 dB in the middle of the audience area, with only about 101 dB of continuous level. That still felt loud for some people. It felt loud for me at times. I wore my earplugs religiously.

To be fair, the PA wasn’t the only thing making noise in the room. The monitor rig and the band’s instrumentation could easily give the total acoustical output a shove that got you into the upper reaches of the 100 dB decade. But even so, you have to realize that 101 dB of continuous system output at room-center resulted from only about 10 watts of continuous input. Remember that I said the limiter for FOH stopped the peaks at 10 dB down. So, that 1000-peak-watt amp was now really only 100 watts maximum, with the continuous power available being 10 dB down from that.

What I’m NOT saying here is that we should all downsize our audio rigs to run on hamster wheels. Headroom (holistic headroom, that is) continues to be a very good idea. There are situations where very large peak-to-continuous ratios have to be handled. What I am saying on balance, though, is that dumping a ton of resources into system capacity that’s actually excess isn’t something I can advise. I just can’t escape this ever-building perception that what a good number of live-music audiences really want are balanced mixes which stay well under an A-weighted level of 100 dB SPL continuous. Add the subwoofer information and you might get to 100 dB or more on another weighting, but that’s a different story.

(And, of course, we have to do what we have to do. Keeping up with a band that’s running hot is a necessity. There were plenty of Fats gigs where I started opening the limiters a little. There was one night where I had to adjust my threshold up to the point where the main amp would show clipping – and then drive hard into that limiting point.)

But there are plenty of gigs that aren’t a slugging match. In those cases, 10 watts of continuous input power might be all that’s actually used. Maybe even less than that. Ten watts can be “too loud” sometimes. I’ve gotten complained at during acoustic shows that people could easily talk over, for goodness sake. I did a few nights at a place with a very nice install that you could barely use in any meaningful way; You would just start pushing some clarity past the monitor wash, and somebody would comment that the music was too loud.

A lot of us aspire to “the big rig,” and I don’t think there’s anything wrong with that on the surface. I simply urge caution. A huge system can be hard to get people to pay for, requires a lot of logistical work, and may be a tremendous amount of excess capacity that never gets leveraged.


The Great, Quantitative, Live-Mic Shootout

A tool to help figure out what (inexpensive) mic to buy.

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.

See that link up there in the header?

It takes you to The Great, Quantitative, Live-Mic Shootout, just like this link does. (Courtesy of the Department of Redundancy Department.)

And that’s a big deal, because I’ve been thinking and dreaming about doing that very research project for the past four years. Yup! The Small Venue Survivalist is four years old now. Thanks to my Patreon supporters, past and present, for helping to make this idea a reality.

I invite you to go over and take a look.


The Grand Experiment

A plan for an objective comparison of the SM58 to various other “live sound” microphones.

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.

Purpose And Explanation

Ever since The Small Venue Survivalist became a reality, I have wanted to do a big experiment. I’ve been itching to round up a bunch of microphones that can be purchased for either below, or slightly above the price point of the SM58, and then to objectively compare them to an SM58. (The Shure SM58 continues to be an industry standard microphone that is recognized and accepted everywhere as a sound-reinforcement tool.)

The key word above is “objectively.” Finding subjective microphone comparisons isn’t too hard. Sweetwater just put together (in 2017) a massive studio-mic shootout, and it was subjective. That is, the measurement data is audio files that you must listen to. This isn’t a bad thing, and it makes sense for studio mics – what matters most is how the mic sounds to you. Listening tests are everywhere, and they have their place.

In live audio, though, the mic’s sound is only one factor amongst many important variables. Further, these variables can be quantified. Resistance to mechanically-induced noise can be expressed as a decibel number. So can resistance to wind noise. So can feedback rejection. Knowing how different transducers stack up to one another is critical for making good purchasing decisions, and yet this kind of quantitative information just doesn’t seem to be available.

So, it seems that some attempt at compiling such measurements might be helpful.

Planned Experimental Procedure

Measure Proximity Effect

1) Generate a 100Hz tone through a loudspeaker at a repeatable SPL.

2) Place the microphone such that it is pointed directly at the center of the driver producing the tone. The front of the grill should be 6 inches from the loudspeaker baffle.

3) Establish an input level from the microphone, and note the value.

4) Without changing the orientation of the microphone relative to the driver, move the microphone to a point where the front of the grill is 1 inch from the loudspeaker baffle.

5) Note the difference in the input level, relative to the level obtained in step 3.

Assumptions: Microphones with greater resistance to proximity effect will exhibit a smaller level differential. Greater proximity effect resistance is considered desirable.

Establish “Equivalent Gain” For Further Testing

1) Place a monitor loudspeaker on the floor, and position the microphone on a tripod stand. The stand leg nearest the monitor should be at a repeatable distance, at least 1 foot from the monitor enclosure.

2) Set the height of the microphone stand to a repeatable position that would be appropriate for an average-height performer.

3) Changing the height of the microphone as little as possible, point the microphone directly at the center of the monitor.

4) Generate pink-noise through the monitor at a repeatable SPL.

5) Using a meter capable of RMS averaging, establish a -40 dBFS RMS input level.

Measure Mechanical Noise Susceptibility

1) Set the microphone such that it is parallel to the floor.

2) Directly above the point where the microphone grill meets the body, hold a solid, semi-rigid object (like an eraser, or small rubber ball) at a repeatable distance at least 1 inch over the mic.

3) Allow the object to fall and strike the microphone.

4) Note the peak input level created by the strike.

Assumptions: Microphones with greater resistance to mechanically induced noise will exhibit a lower input level. Greater resistance to mechanically induced noise is considered desirable.

Measure Wind Noise Susceptibility

1) Position the microphone on the stand such that it is parallel to the floor.

2) Place a small fan (or other source of airflow which has repeatable windspeed and air displacement volume) 6 inches from the mic’s grill.

3) Activate the fan for 10 seconds. Note the peak input level created.

Assumptions: Microphones with greater resistance to wind noise will exhibit a lower input level. Greater resistance to wind noise is considered desirable.

Measure Feedback Resistance

1) Set the microphone in a working position. For cardioid mics, the rear of the microphone should be pointed directly at the monitor. For supercardioid and hypercardioid mics, the the microphone should be parallel with the floor.

2a) SM58 ONLY: Set a send level to the monitor that is just below noticeable ringing/ feedback.

2b) Use the send level determined in 2a to create loop-gain for the microphone.

3) Set a delay of 1000ms to the monitor.

4) Begin a recording of the mic’s output.

5) Generate a 500ms burst of pink-noise through the monitor. Allow the delayed feedback loop to sound several times.

6) Stop the recording, and make note of the peak level of the first repeat of the loop.

Assumptions: Microphones with greater feedback resistance will exhibit a lower input level on the first repeat. Greater feedback resistance is considered desirable.

Measure Cupping Resistance

1) Mute the send from the microphone to the monitor.

2) Obtain a frequency magnitude measurement of the microphone in the working position, using the monitor as the test audio source.

3) Place a hand around as much of the mic’s windscreen as is possible.

4) Re-run the frequency magnitude measurement.

5) On the “cupped” measurement, note the difference between the highest response peak, and that frequency’s level on the normal measurement.

Assumptions: Microphones with greater cupping resistance will exhibit a smaller level differential between the highest peak of the cupped response and that frequency’s magnitude on the normal trace. Greater cupping resistance is considered desirable.


Measuring A Cupped Mic

What you might think would happen isn’t what happens.

Please Remember:

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

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

The most popular article on this site to date is the one where I talk about why cupping a vocal mic is generally a “bad things category” sort of experience. In that piece, I explain some general issues with wrapping one’s hand around a microphone grill, but there’s something I didn’t do:

I didn’t measure anything.

That reality finally dawned on me, so I decided to do a quck-n-dirty experiment on how a microphone’s transfer function changes when cupping comes into play. Different mics will do different things, so any measurement is only valid for one mic in one situation. However, even if the results can’t truly be generalized, they are illuminating.

In the following picture, the red trace is a mic pointing away from a speaker, as you would want to happen in monitor-world. The black trace is the mic in the same position, except with my hand covering a large portion of the windscreen mesh.

You would think that covering a large part of the mic’s business-end would kill off a lot of midrange and high-frequency information, but the measurement says otherwise. The high-mid and HF information is actually rather hotter, with large peaks at 1800 Hz, 3900 Hz, and 9000 Hz. The low frequency response below 200 Hz is also given a small kick in the pants. Overall, the microphone transfer function is “wild,” with more pronounced differences between peaks and dips.

The upshot? The transducer’s feedback characteristics get harder to manage, and the sonic characteristics of the unit begin to favor the most annoying parts of the audible spectrum.

Like I said, this experiment is only valid for one mic (a Sennheiser e822s that I had handy). At the same time, my experience is that other mics have “cupping behavior” which is not entirely dissimilar.


Thoughts On Earplugs

They’re a good idea, and you don’t have to spend much to get good ones.

Please Remember:

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

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

The Video

The Summary

You only get one pair of ears, so protect them with plugs. Don’t let anyone tell you not to do so. “Flat response” plugs can be both generic or custom fitted, with custom molds having a large advantage in overall comfort.


You Don’t Just Need A Bigger Amp

Headroom is a holistic thing. If you run out of it in one place, getting more of it somewhere else isn’t enough.

Please Remember:

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

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

The Video

The Script

Let’s say that an audio human has a mixing console that’s feeding a loudspeaker system. (That makes sense, right? Most of us do that a lot.) This loudspeaker system is nifty because it’s magic. It never clips. The only limit on output is how much voltage the console can deliver.

The weird bit is that the console can’t really swing much voltage at its outputs. It clips at a peak of 1 volt. Another weird thing is that the console doesn’t have any inboard mic pres. Those are separate. (I know this is really strange, but I’m trying to make a point.)

Our aforementioned audio human just happens to have a mic pre that also reaches its maximum output at a 1 volt peak. They connect a signal source to the preamp, crank the pre until it’s just barely under clipping, set a fader at 0 dB, and…it’s not loud enough.

So, what do they do?

If you said, “They need a console with more output capability,” you’re exactly right.

It wouldn’t make any sense to buy a mic pre with more output, right? If the console output clips at 1 volt, what good does it do to have a mic pre that will deliver 12 volts into the console? You can drive the signal to the mix outputs a lot harder, but all that gets you is more distortion.

Obtaining and connecting an upstream device with more output is kinda absurd, frankly. It’s not a solution at all. The console output is the limiting factor.

But here’s the thing.

People take this actually non-sensical approach with amps and speakers all the time. Some of the confusion is understandable. Amplifier and speaker power ratings aren’t necessarily intuitive, for one, and passive speakers don’t have level meters as a rule. There’s also all the complexity involved with trying to describe the limits of a multi-device speaker enclosure with a single number.

I get that there’s nuance involved here.

But here’s the thing. Speakers, like everything else, have a maximum undistorted output point. It’s a peak level – a point beyond which there is no more “instantaneous” sound pressure to be had. If you have, say, a loudspeaker that can handle a peak input of 1000 watts, and an amplifier that can put a 1000 watt peak input through that box…you’re there. Your system is maximized. Any more available amplifier power is wasted on both driver distortion AND the chance that you might wreck your speaker.

But people see those nasty little clip lights on their amps, and think: “Gosh, I need to buy a more powerful amplifier!” They get obsessed with headroom, but in this compartmentalized way that only involves the amp.

Actually, unless the amp’s peak (NOT CONTINUOUS – PEAK)…unless the amp’s peak output is half (or less) of the speaker’s maximum peak rating, you do NOT need a larger amp. Getting a more powerful amplifier only gets you more headroom in the amplifier, when what you actually want is more headroom throughout the entire system output section.

What you need is a speaker that gets louder for the same amount of input. Or a bigger amplifier AND a speaker that can handle the additional power. Or just “more PA” in general – although that’s a whole other can of worms in itself.


Why Are Faders Labeled Like That?

Gain multipliers are hard to read.

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.

I’ve done a lot of typing on this site, and I’m worried that it’s getting stale – so, how about some video?


Drivers Don’t Have To Die With A Bang

Sane powering shields you from accidents.

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.

I once lived in abject terror of pops, clicks, and bangs. I was once frightened by the thought of a musician unplugging their instrument from a “hot” input before I found the mute button. This was a result of my early experience in audio, where well-meaning (but incorrect) people had assured me that such noises were devastating to loudspeakers. A good solid “thump” from powering up a console when the amps were already on, and some poor driver would either:

A) Take another step towards doom, or…

B) Blow up like that one space station that could be confused with a small moon.

Well, that’s just a load of horsefeathers, but like all audio myths, a kernel of truth can be found. The kernel of truth is that loudspeakers CAN be destroyed by a large spike of input. There’s a reason that drivers and loudspeaker enclosures have peak ratings. Those are “Do Not Exceed” lines that you are smart to avoid crossing. Here’s the deal, though – if you’re using a sane powering strategy with passive boxes, or are using any truly decent powered speaker, worry is essentially unnecessary.

An amplifier simply can not “swing” more voltage than is available from the supply. If the peak voltage available from the amp results in power dissipation equal to or less than what the loudspeaker can handle, a brief transient won’t cook your gear. The instantaneous maximum power will be in the safe range, and the whole signal won’t last long enough for the continuous power to become a factor. An active box that’s well designed will either be powered in such a way, or it may be overpowered and then limited back into a safe range.

So, when a system is set up correctly, the odd mishap isn’t necessarily dangerous. It’s just displeasing to hear.

I believe that the persistence of this myth is due to folks who get talked into “squeezing maximum performance” out of their loudspeakers. They’re told that they have to use very large amplifiers to drive the boxes they have, and so that’s what they do. They hook up amps that can handily deliver power far beyond the “Do Not Exceed” line specified by peak ratings. If they take no other safety precautions, they ARE playing with fire. One good, solid accident, and that may be it for a driver. (If I might be so bold, I would recommend that those folks instead use my speaker powering strategy instead of “spend lots more, maybe get a touch louder, and hope you’re lucky.”)

The worrier doesn’t have to be you. Keep things reasonable, and you’ll be very unlikely to lose money because somebody yanked a cable.