Tag Archives: Bang For The Buck

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 3 feet 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 -20 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) 6 inches 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 four times.

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

Assumptions: Microphones with greater feedback resistance will exhibit a lower input level on the fourth 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.


Actually, Your Equipment Is Probably Fine

Working as a team is more important than most anything.

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 is from another article that I wrote for Schwilly Family Musicians: “What they had failed to do was to play as a team, and that made their perfectly adequate gear SEEM like a problem area.”

Read the whole thing for free, here.


Console Envy

When it comes to sound quality, any console capable of doing the show will probably be fine.

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

Which console sounds best? The one with the features you need. If an inexpensive mixer has all the necessary features for your shows, spending more doesn’t have much of a point.


Baskets, Bees, and Flies

Quality generally beats quantity.

Please Remember:

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

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

Sometimes, more IS more. It doesn’t matter how nice your mic cables are if you don’t have enough of them. If the show absolutely requires 24 channels, and you have a console with 16 really amazing channels…well, you’re still short by eight.

Yet, there are still plenty of instances where “a handful of bees is better than a basket of flies” (as Moroccans might say).

For instance, some folks are really hung up on the idea that a “main” PA speaker should be built around a 15″-diameter low-frequency driver. The idea is that bigger is better, but that’s not always so. Given a choice, I’ll take a good box built around a 12″ cone over a mediocre offering constructed around a 15. A well-designed 12 can be kinder to the vocals, because the cone driver is better at “playing” higher and covering the range that a small horn-driver can’t quite reach down into. Sure, the 12 probably won’t go as low, but if you want to be “loud” below 100 Hz you’re going to want subwoofers anyway. (For the record, I would never turn my nose up at a perfectly decent box that used a 15 or two.)

Also talking about speakers, there are people who believe a PA with more boxes is superior to a rig with fewer. The problem is that you have to take deployment into account. If you already have the necessary horizontal and vertical coverage happening, more boxes just act to cause more interference problems. The system looks cool because it’s bigger, and it gets louder because there are more boxes, but it doesn’t actually sound better. It might even sound terrible with all that comb-filtering going on. Coverage is sort of like what The Mad Hatter said to Alice: “When you get to the end, stop.”

This applies to bands too, especially when it comes to vocalists. One really brilliant singer with one mic is almost always light-years better than a whole group of vocalists of questionable quality. Beyond the basic aesthetics, not-so-hot singers tend to require a lot more gain to be heard (because they usually haven’t developed much vocal power), and that can easily lead to a system being run on the knife-edge of feedback all night.

…and speaking of people, how about crew-members? Any day of the week, and twice on Sunday, I’ll gladly take one knowledgeable, pleasant, and punctual helper over 15 punters who are late, surly, and have no idea what’s going on.

Tossing more and more junk at a problem rarely fixes the problem. You might eventually smother your issue or manage to distract from it, but the bugbear is still sitting beneath the pile. Applying a sufficient fix, on the other hand, works very reliably. There are times when you need “more.” There’s no getting around that. However, it’s important to avoid using “more” as a substitute for having what will actually do the job effectively.


Single-Ended Measurement

I really prefer it over minutes on-end of loud pink-noise.

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.

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

Today, I helped teach a live-sound class at Broadview Entertainment Arts University. We put a stage together, ran power, and set both Front Of House (FOH) and monitor-world loudspeakers. To cap-off the day, I decided to show the students a bit about measuring and tuning the boxes that we had just finished squaring away.

The software we used was Room EQ Wizard.

The more I use “REW,” the more I like the way it works. Its particular mode of operation isn’t exactly industry-standard, but I do have a tendency to ignore the trends when they aren’t really helpful or interesting to me. Rather than continually blasting pink-noise (statistically uncorrelated audio signals with equal energy per octave from 20 Hz to 20 kHz) into a room for several minutes while you tweak your EQ, Room EQ Wizard plays a predetermined sine-sweep. It then shows you a graph, you make your tweaks based on the graph, you re-measure, and iterate as many times as needed.

I prefer this workflow for more than one reason.

Single Ended Measurements Are Harder To Screw Up

The industry-standard method for measuring and tuning loudspeakers is that of the dual-FFT. If you’ve used or heard of SysTune or SMAART, among others, those are dual-FFT systems. You run an essentially arbitrary signal through your rig, with that signal not necessarily being “known” ahead of time. That signal has to be captured at two points:

1) Before it enters the signal chain you actually want to test.

2) After it exits the signal chain in question.

And, of course, you have to compensate for any propagation delay between those two points. Otherwise, your measurement will get contaminated with statistical “noise,” and become harder to read in a useful way – especially if phase matters to you. Averaging does help with this, to be fair, and I do average my “REW” curves to make them easier to parse. Anybody who has taken and examined a measurement trace in a real room knows that unsmoothed results look pretty terrifying.

In any event, dual-FFT measurements tend to be more difficult to set up and run effectively. On top of how easy it is to screw up ANY measurement, whether by measuring the wrong thing, forgetting an upstream EQ, or putting the mic in a terrible spot, you have the added hassles of getting your two measurement points routed and delay-compensated.

Over the years, dual-FFT packages have gotten much better at guiding users through the process, internally looping back the reference signal, and automatically picking compensation delay times. Even so, automating a complicated process doesn’t make the process less complicated. It just shields you from the complexity for as long as the automation can help you. (I’m not bagging on SMAART and SysTune here. They’re good bits of software that plenty of folks use successfully. I’m just pointing some things out.)

Single Ended, “Sweep” Measurements Can Be Quieter (And Less Annoying)

Another issue with measurements involving broadband signals is that they have greater susceptibility to in-room noise. As a whole, the noise may be quite loud. However, any given frequency can’t be running very “hot,” as the entire signal has to make it cleanly through the signal path. As such, noise in the room easily contaminates the test at the frequencies contained within that noise, unless you run the test signal loudly enough. With a single-ended, sine-sweep measurement, the instant that the measurement tone is at a certain frequency, the entire system output is dedicated to that frequency alone. As such, if you have in-room noise of 50 dB SPL at 1 kHz, running your measurement signal at 70 dB SPL should completely blow past the noise – while remaining comfortable to hear. With broadband noise, the measurement signal in the same situation might have to be 90 dB SPL.

Please note that single-ended measurements of broadband signals DO exist, and they have similar problems with noise as compared to broadband-noise, dual-FFT solutions.

The other nice thing about “sweep” measurements is that everybody gets a break from the noise. For 10 seconds or so, a rising tone sounds through the system, and then it stops. This is a stark contrast to minutes of “KSSSSSHHHHHH” that otherwise have to be endured.

Quality, Single Ended Measurement Software Can Be Cheaper

A person could conceivably design and build single-ended measurement software, and then sell it for a large amount of money. A person could also create dual-FFT software and give it away for free (Visual Analyzer is a good example).

However, on average, it seems that when it comes time to bring “easy to use” and “affordable” together, single-ended is where you’ll have to look. I really like Visual Analyzer, but you really, really have to know what you’re doing to use it effectively. SMAART and SysTune are user-friendly while also being incredibly powerful, but cost $700 – $1000 to acquire.

Room EQ Wizard is friendly (at least to me), and free. It’s hard to beat free when it’s also good.


I want to be careful to say (again) that I’m not trying to get people away from the highly-developed and widely accepted toolsets available in dual-FFT measurement packages. What I’m trying to say is that “dual-FFT with broadband noise in pseudo-realtime” isn’t the only way to measure and tune a sound system. There are other options that are easier to get into, and you can always step up later.


How To Buy A Microphone For Live Performance

A guest-post for Schwilly Family Musicians

Please Remember:

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

vintage_microphone-wallpaper-1280x800

From the article: “At the same time, though, a LOT of mics that are great for recording are a giant ball of trouble for live audio. Sure, they sound perfect when you’re in a vocal booth with headphones on, but that’s at least one whole universe removed from the brutal world of concert sound. They’re too fragile, too finicky, too heavy, their pickup patterns are too wide, and you can’t get close enough to them to leverage your vocal power.”


The whole thing is available for free, so go ahead and take a gander.


How Much Light For Your Dollar?

Measurements and observations regarding a handful of relatively inexpensive LED PARs.

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.

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

I’m in the process of getting ready for a pretty special show. The album “Clarity” by Sons Of Nothing is turning 10, and a number of us are trying to put together one smasher of a party.

Of course, that means video.

And our master of all things videographic is concerned about having enough light. We can’t have anybody in the band who’s permanently stuck in “shadow.” You only get one chance to shoot a 10th anniversary concert, and we want to get it right.

As such, I’m looking at how to beef up my available lighting instruments. It’s been a long while since I’ve truly gone shopping for that old mainstay of small-venue lighting, the LED wash PAR, but I do take a look around every so often. There’s a lot to see, and most of it isn’t very well documented. Lighting manufacturers love to tell you how many diodes are in a luminaire, and they also like to tell you how much power the thing consumes, but there appears to be something of an allergy to coughing up output numbers.

Lux, that is. Lumens per square meter. The actual effectiveness of a light at…you know…LIGHTING things.

So, I thought to myself, “Self, wouldn’t it be interesting to buy some inexpensive lights and make an attempt at some objective measurement?”

I agreed with myself. I especially agreed because Android 4.4 devices can run a cool little Google App called “Science Journal.” The software translates the output from a phone’s ambient light sensor into units of lux. For free (plus the cost of the phone, of course). Neat!

I got onto Amazon, found myself a lighting brand (GBGS) that had numerous fixtures available for fulfillment by Amazon, and spent a few dollars. The reason for choosing fulfillment from Amazon basically comes down to this: I wanted to avoid dealing with an unknown in terms of shipping time. Small vendors can sometimes take a while to pack and ship an order. Amazon, on the other hand, is fast.

The Experiment

Step 1: Find a hallway that can be made as dark as possible – ideally, dark enough that a light meter registers 0 lux.

Step 2: At one end, put the light meter on a stand. (A mic stand with a friction clip is actually pretty good at holding a smartphone, by the way.)

Step 3: At the other end, situate a lighting stand with the “fixture under test” clamped firmly to that stand.

Step 4: Measure the distance from the lighting stand to the light meter position. (In my case, the distance was 19 feet.)

Step 5: Darken the hallway.

Step 6: Set the fixture under test to maximum output using a DMX controller.

Step 7: Allow the fixture to operate at full power for roughly 10 minutes, in case light output is reduced as the fixture’s heat increases.

Step 8: Ensure the fixture under test is aimed directly at the light meter.

Step 9: Note the value indicated by the meter.

Important Notes

A relatively long distance between the light and the meter is recommended. This is so that any positioning variance introduced by placing and replacing either the lights or the meter has a reduced effect. At close range, a small variance in distance can skew a measurement noticeably. At longer distances, that same variance value has almost no effect. A four-inch length difference at 19 feet is about a 2% error, whereas that same length differential at 3 feet is an 11% error.

It’s important to note that the hallway used for the measurement had white walls. This may have pushed the readings higher, as – similarly to audio – energy that would otherwise be lost to absorption is re-emitted and potentially measurable.

It was somewhat difficult to get a “steady” measurement using the phone as a meter. As such, I have estimated lux readings that are slightly lower than the peak numbers I observed.

These fixtures may or may not be suitable for your application. These tests cannot meaningfully speak to durability, reliability, acceptability in a given setting, and so on.

The calculation for 1 meter lux was as follows:

19′ = 5.7912 m

5.7912 = 2^2.53 (2.53 doublings of distance from 1m)

Assumed inverse square law for intensity; For each doubling of distance, intensity quadruples.

Multiply 19′ lux by 4^2.53 (33.53)

Calculated 1 meter lux values are just that – calculated. LED PAR lights are not a point-source of light, and so do not behave like one. It requires a certain distance from the fixture for all the emitters to combine and appear as though they are a single source of light.

The Data

The data display requires Javascript to work. I’m sorry about that – I personally dislike it when sites can’t display content without Javascript. However, for the moment I’m backed into a corner by the way that WordPress works with PHP, so Javascript it is.


Ascending sort by:


The Pros And Cons Of Distributed Monitor Mixing

It’s very neat when it works, but it’s not all sunshine, lollipops, and rainbows.

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.

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

Along with folks who rock the bars and clubs, I also work with musicians who rock for church. Just a few months ago, as City Presbyterian’s worship group was expanding (and needing more help with monitoring), I decided to put the players on a distributed monitor-mix system. What I mean by a “distributed” system is that the mix handling is decentralized. Each musician gets their own mini-mixer, which they use to “run their own show.”

The experience so far has been basically a success, with some minor caveats. The following is a summary of both my direct observations and theoretical musings regarding this particular monitoring solution.


Pro: In-Ear Monitors Become Much Easier For The Engineer

One downside to in-ears is that the isolation tends to require that everyone get a finely tuned mix of many channels. This is especially true when you’re running a quiet stage, where monitor world is required to hear much of anything. What this mandates is a lot of work on behalf of each individual performer, with the workload falling squarely on the shoulders of the audio human.

Distributed monitor mixing takes almost all of the workload off the sound operator, by placing the bulk of the decision making and execution in the hands of individual players. If the lead guitarist wants more backup vocals, they just select the appropriate channel and twist the knob. If they want the tonality of a channel altered, they can futz with it to their heart’s content. Meanwhile, the person driving the console simply continues to work on whatever they were working on, without giving much thought to monitor world.

Con: Monitors Become Harder For The Player

Much like effort and preparation, complexity for the operation of a given system can neither be created nor destroyed. It can only be transferred around. A very, very important thing to remember about distributed monitor mixing is this: You have just taken a great deal of the management and technical complexity involved in mixing monitors, and handed it to someone who may not be prepared for it. Operating a mix-rig in a high-performance, realtime situation is not a trivial task, and it takes a LOT of practice to get good at it. To be sure, a distributed approach simplifies certain things (especially when in-ears essentially delete feedback from the equation), but an inescapable reality is that it also exposes a lot of complexity that the players may have had hidden from them before. Things like sensible gain staging and checking for sane limiter settings are not necessarily instinctual, and may not be a part of a musician’s technical repertoire on the first day.

Also, as the engineer, you can’t just plug in each player’s mixer and mentally check out. You MUST have some concept of how the mixers work, so that you can effectively support your musicians. Read the manual, plug in one of the units, and turn the knobs. Personal mixers may be operated by individual players, but they really are part of the reinforcement rig – and thus, the crew is responsible for at least having some clue about how to wield them.

Pro: You Don’t Necessarily Have To Use In-Ears

I have yet to encounter a personal-mix system that didn’t include some sort of “plain vanilla” line output. If the musicians want to drive a powered wedge (or an amplifier for a passive wedge) with their mixer, they can.

Con: Not Using In-Ears May Cause Trouble

As I said before, mixing in a high-performance situation isn’t an easy thing that humans are naturally prepared to do. Life gets even more hairy in a “closed-loop” situation – i.e., onstage monitoring with mics and loudspeakers. A musician may dial their piece of monitor world (at a bare minimum) into SCREAMING feedback without realizing their danger. They may not recognize how to get themselves out of the conundrum.

And, depending on how your system works, the audio human may not be able to “right the ship” from the mix position.

Even if they don’t get themselves swallowed by a feedback monster, a player can also run their mix so loud that they’re drowning everybody else, including the Front Of House mix…

Pro: Integrated Ecosystems Are Powerful And Easy

As more digital console “ecosystems” come online, adding distributed mixing is becoming incredibly easy. For instance, Behringer’s digital Powerplay products plug right into Ultranet with almost zero fuss. If your console has Ultranet built-in, you don’t have to worry about tapping inserts or direct outs. You just run a Cat5/ Cat6 cable to a distribution module, the module sends data and power over the other Cat5/6 runs, and everything just tends to work.

Con: Once You’ve Picked Your Ecosystem, You’ll Have To Stay There

Integrated digital audio ecosystems make things easy, but they tend to only play nice within the same extended family of products. You can’t run an Ultranet product on an Aviom monitor-distro network, for instance. More universal options do exist, but the universality tends to come with a large price premium. Whenever you go a certain way with a system of personal mixers, you’re making a big commitment. The jump to a different product family may be difficult to do…or just a flat-out expensive replacement, depending upon the system flexibility.

Pro: Everybody Can Have Their Own Mixer

Distributed mixing can be a way to banish all monitor-mix sharing for good. Everybody in the band can not only have their own mix, but their own channel equalization as well. If the guitar player wants the bass to sound one way, and the bass player wants the bass to sound totally different, that option is now very viable. Each musician can build intricate presets inside their own piece of hardware, without necessarily having to consult with anyone else.

Con: Everybody Having Their Own Mixer Is Expensive

Expensive is a relative term, of course. With a Powerplay system, outfitting a five-piece band is about as expensive as buying a couple-three “pretty dang nice,” powered monitor wedges. Other systems involve a lot more money, however. Also, even with an affordable product-line, adding a new member to the band means the expense of adding another personal mixer and attendant accessories.

Pro: Personal Mixing Is Luxurious

When we deployed our distributed system, one of the comments I got was “This is what we’ve always wanted, but couldn’t have. It should always have worked this way.” Everybody getting their own personal, instantly customizable mix is a “big league” sort of setup that is now firmly within reach for almost any band. Under the right circumstances, moving the on-deck show into the right place can transform from a slog to a joy.

Con: Not Everybody May Buy In To The Idea

The adoption of a distributed monitor mixing system is like all personal monitoring: Personal. The problem is that you have to try it to find out if you want to deal with it or not. Unless someone categorically states at the outset that they want no part of individualized mixing, the money has to be spent to let them give it a whirl.

…and they may decide that it’s just not for them, with only 30 minutes of use on their mixer and the money already spent. You just have to be ready for this, and be prepared to treat it as a natural cost of the system. Forcing someone to use a monitoring solution that they dislike is highly counterproductive.

Distributed monitor mixing, like all live-audio solutions, is neither magic nor a panacea. It may be exactly the right choice for you, or it may be a terrible one. As with everything else, there’s homework to be done, and nobody can do it but you. One size does not fit all.


Should You Go To Audio School?

I went, and I loved it, but I don’t universally recommend it.

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.

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

I’m not entirely sure if me being a graduate of The Conservatory Of Recording Arts And Sciences reflects well or poorly on the institution. I definitely did NOT walk out of there and summarily change the world, but I have made plenty of friends and mixed a whole bunch of shows that were well received.

In any case…

I went to The Conservatory. I loved it. It was the best academic experience of my entire life. You would think, then, that I would wholeheartedly recommend to anyone considering a run in this business that they also go to school for the craft.

That’s not the case, though.


At the level of the general population, we are slowly waking up to the reality that the “diploma in hand” is really not a golden ticket. Our collective, aggressive somnolence in regards to this realization can be partially excused; For a long time, school was THE key to the brighter future. There was quite a long count of years where the piece-o-paper did indeed function well as a gate pass to getting a gig. There are some professions that still absolutely require proof of getting through the coursework to even get started. For many occupations, though, successful passage through related education is now a pretty mediocre commodity. You went to school? So? The 10,000 other people who want to do this job also did.

Having the education on your resume is no longer anything that makes you stand out from the pack. It’s not at all rare.

But higher-ed institutions of all types, especially those that really need your tuition dollars, won’t tell you that. They live on your believing that the best way to get into [insert profession here] is to have some sort of diploma. Like I said, though, the diploma no longer marks you as exceptional. It just shows that you were able to spend enough money and hang on long enough to get your credit hours.


From the above, you might think that I’m against school. I’m not. I’m against believing that school is something that it inherently isn’t.

School is not, at its core, an entry card into a profession or socio-economic group. It can act as those things under certain circumstances, but that’s not inherently what school is.

School is actually your becoming familiar with basic concepts and vocabulary such that you have a chance to understand your real education, which is the doing of the work in real life. It’s the mental foundation for asking the really interesting questions, questions that tend not to be covered in school.

(There are educational institutions which get into those questions, but they do so only at the very highest levels. Original research, the prime-example of this, is not school. It’s “doing the work in real life,” just in an academic setting where the goals are more than making a profit this month.)

The point of school is to make you able to learn something later, when you’re not in the classroom, lab, or other controlled environment.

So, if that’s the premise I’m going with, why would I NOT encourage you (like crazy) to go to school for sound? Doesn’t recording or live-audio school give you a crucial foundation for a future life in noise-louderization and electron inconveniencing?

Well, it can, but it’s not the only way to get there.


I went to recording school at just around the turn of the century. Digital consoles were out there, but were still a revolutionary concept for a lot of us in the classroom. The music industry still revolved around rock bands being recorded in big, expensive rooms through big, expensive consoles, connected to big, expensive outboard gear. CPU-based audio workstations were just at the doorway of competing with Pro Tools rigs running DSP cards. The project-studio revolution was definitely in full swing, but audio was still in a place where you could spend a lot without getting a lot.

You also have to realize that the Internet was in the midst of revolutionizing everything, but not nearly as far along as it is now. Information that’s easy to find these days was still difficult to ferret out then. YouTube, and a million people doing “how to hook up your sound system” did not exist. Not everybody posted their manuals and free(!) editor software online.

What audio schools had at that time was full-featured gear, actual studio rooms like what were in vogue, information, and the opportunity to do “lab” work that combined all that. They could charge you a fair amount for the privilege, and be basically justified in doing so. They were riding that bleeding edge of a business that traditionally worked on the “master and apprentice” model anyway, but had become big enough for commoditized education to handle the basics.

Do you know what’s changed since then?

The schools have newer gear.

They charge quite a bit more for tuition.

Gear with immense functionality has dropped in price.

All the information you need is available almost instantaneously, often for free.

Huge sections of the music business have stopped being “big industry,” and have returned to their DIY, “punk rock” roots.

What hasn’t changed at all is that “hands-on” time is still the most precious part of learning the craft.

To be brutally frank, as far as I can tell, for the price of an audio school program you can buy your own gear that – while certainly not top-shelf – will have all the features necessary for you to learn much more than the bare basics. Once you get comfortable with signal flow fundamentals, you could then start looking for bands to work with, and maybe even make some money while you establish experience. A diploma is worth very little compared to real experience, a reputation, and having some of your own equipment.


None of this is to write off academic audio programs entirely. If you truly want to go to school for sound, you should – but I would encourage you to look at non-traditional factors when choosing a school. Forget about the nameplates on the gear and the manufacturer-sponsored certification programs. Forget about whether or not the live-sound lab has the biggest and loudest flown array ever assembled. Forget about the stories of (a very small minority of their students, probably) who are working with giant artists and getting their names on industry awards that are mostly based on sales. Rather, think about:

How much hands-on time is part of the curriculum? The more there is, the better.

Related to the above, how much real, honest-to-goodness portfolio material will you have when you graduate? The more you can get, the better.

How much recruiting is done by potential employers at the school? Do local production companies go looking for graduates? The more of that there is, the better.

Will there be easy opportunities to meet and form relationships with people working at local, regional, and national levels? The more of those, the better.

We’re in a new age where the traditional barriers to entry are nearly nonexistent. If you’re going to go to school, go to a place that serves as a functional launchpad for your career, not merely a factory for people who can answer questions on tests. To use the language of Seth Godin, look for a place that prepares you to pick yourself, rather than for other people to pick you. If I had all of this to do over again, and I went to school, I would go to a school like that.

Heck, I want to teach there.


Double Hung Discussion

It’s not magic, and it may not be for you. It works for me, 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.

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

On the heels of my last article, it came to may attention that some folks were – shall we say – perplexed about my whole “double hung” PA deployment. As can be the case, I didn’t really go into any nuance about why I did it, or what I expected to get out of it. This lead some folks to feel that it was a really bizarre way to go about things, especially when a simpler solution might have been a better option.

The observations I became aware of are appropriate and astute, so I think it’s worth talking about them.

Why Do It At All?

First, we can start with that logistics thing again.

When I put my current system together, I had to consider what I was wanting to do. My desire was to have a compact, modular, and flexible rig that could “degrade gracefully” in the event of a problem. I also had no desire to compete with the varsity-level concert systems around town. To do so would have required an enormous investment in both gear and transport, one that I was unwilling (and unable) to make.

What I’ve ended up with, then, is a number of smaller boxes. If I need more raw output, I can arrange them so that they’re all hitting the same general area. I also have the option of deploying for a much wider area, but with reduced total output capability. I wouldn’t have that same set of options with a small number of larger, louder enclosures.

That’s the basic force behind why I have the rig that I have. Next come the more direct and immediate issues.

The first thing is just a practical consideration: Because my transport vehicle isn’t particularly large, I don’t really have the necessary packing options required to “leave gear on the truck.” If I’m getting the rig out, I might as well get all of it out. This leads to a situation where I figure that I might as well find a way to deploy everything all the time. The gear is meant to make noise, not sit around. “Double hung” lets me do that in a way that makes theoretical sense (I’ll say more on why in a bit).

The second reason is less practical. I have a bit of a penchant for the unconventional and off-the-wall. I sometimes enjoy experiments for the sake of doing them, and running a double hung system is just that kind of thing. I like doing it to find out what it’s like to do it.

Running double hung is NOT, by any means, more practical than other deployments. Especially if you’re new to this whole noise-louderization job, going with this setup is NOT some sort of magical band-aid that is going to fix your sound problems. Also, if you’re getting good results with a much simpler way of doing things, going to the extra trouble very well may not be worth it.

At the same time, though, the reality of making this kind of deployment happen is not really all that complicated. You can do it very easily by connecting one pair to the left side of your main mix, and the other pair to the right side. Then, you just pan to one side or the other as you desire.

System Output And Response

Up above, I mentioned that running my system as a double hung made sense in terms of audio theory. Here’s the explanation as to why. It’s a bit involved, but stick with me.

I haven’t actually measured the maximum output of my FOH mid-highs, but Turbosound claims that they’ll each make a 128 dB SPL peak. I’m assuming that’s at 1 meter, and an instantaneous value. As such, my best guess at their maximum continuous performance, run hard into their limiters, would be 118 dB SPL at 1 meter.

If I run them all together as one large rig, most people will probably NOT hear the various boxes sum coherently. So, the incoherent SPL addition formula is what’s necessary: 10 Log10[10^(dB SPL/ 10) + 10^(dB SPL/ 10)…]. What I put into Wolfram Alpha is 10 Log10[10^11.8 + 10^11.8 + 10^11.8 + 10^11.8].

What I get out is a theoretical, total continuous system output of 124 dB SPL at 1 meter, ignoring any contribution from the subwoofers.

At this point, you would be quite right to say that I can supposedly get to that number in one of two ways. The first, simple way, is to just put everything into all four boxes. The second, not simple way is to put some things in some boxes and not in others. Either way, the total summed sound pressure should be basically the same. The math doesn’t care about the per-box content. So, why not just do it simply?

Because there’s more to life than just simply getting to the maximum system output level.

By necessity of there being physical space required for the speakers to occupy, the outer pair of enclosures simply can’t create a signal that arrives at precisely the same moment as the signal from the inner pair, as far as the majority of the audience can perceive. Placed close together, the path-length differential between an inner box and an outer box is about 0.0762 meters, or 3 inches.

That doesn’t seem so bad. The speed of sound is about 343 meters/ second in air, so 0.0762 meters is 0.22 ms of delay. That also doesn’t seem so bad…

…until you realize that 0.22 ms is the 1/2 cycle time of 2272 Hz. With the outer boxes being 1/2 cycle late, 2272 Hz would null (as would other frequencies with the same phase relationship). If everything started as measuring perfectly flat, introducing that timing difference into a rig with multiple boxes producing the same material would result in this transfer function:

combfiltering0.22ms

Of course, everything does NOT start out as being perfectly flat, so that craziness is added onto whatever other craziness is already occurring. For most of the audience, plenty of phase weirdness is going on from any PA deployed as two, spaced “stacks” anyway. To put it succinctly, running everything everywhere results in even more giant holes being dug into the critical-for-intelligibility range than were there before.

Running double hung, where the different pairs of boxes produce different sounds, prevents the above problem from happening.

So, when I said that I was running double hung for “clarity,” I was not doing it to fix an existing clarity problem. I was preventing a clarity problem from manifesting itself.

Running absolutely everything into every mid-high, and then having all those mid-highs combine is a simple way to make a system’s mid-highs louder. It’s also a recipe for all kinds of weird phase interactions. These interactions can be used intelligently (in an honest-to-goodness line-array, for instance), but for most of us, they actually make life more difficult. Louder is not necessarily better.

More On Output – Enough Rig For The Gig?

For some folks reading my previous installment, there was real concern that I hadn’t brought enough PA. They took a gander at the compactness of the rig, and said, “There’s no way that’s going to get big-time sound throughout that entire park.”

The people with that concern are entirely correct.

But “rock and roll level everywhere” was not at all what I was trying to do.

The Raw Numbers

What I’ve found is that many people do NOT actually want everything to be “rock and roll” loud over every square inch of an event area. What a good number of events actually want is a comfortable volume up close, with an ability to get away from the noise for the folks who aren’t 100% interested. With this being the case, investing in a system that can be clearly heard at a distance of one mile really isn’t worthwhile for me. (Like I said, I’m not trying to compete with a varsity-level sound company.)

Instead, what I do is to deploy a rig that’s in close proximity to the folks who do want to listen, while less interested people are at a distance. Because the folks who want more volume are closer to the PA, the PA doesn’t have to have crushing output overall. For me, the 110 dB SPL neighborhood is plenty loud, and I can do that for the folks nearby – by virtue of them being nearby.

Big systems that have to cover large areas often have the opposite situation to deal with: The distance differential between the front row and the back row can actually be smaller, although the front row is farther away from the stacks in an absolute sense. With my rig, the people up close are probably about three meters from the PA. The folks far away (who, again, aren’t really interested) might be 50 meters away. That’s more than a 16-fold difference. At a bigger show, there might be a barricade that’s 10 meters from the PA, with the main audience extending out to 100 meters. That’s a much bigger potential audience, but the difference in path lengths to the PA is only 10-fold.

Assuming that the apparent level of the show drops 6 dB for every doubling of distance, my small show loses about 24 decibels from the front row to the folks milling around at 50 meters. The big show, on the other hand, loses about 20 dB. (But they have to “start” much louder.)

That is, where the rubber hits the road is how much output each rig needs at 1 meter. At the big show, they might want to put 120 dB SPL into the front seats. To do that, the level at 1 meter has to be 140 dB. That takes a big, powerful PA. The folks in the back are getting 100 dB, assuming that delays aren’t coming into the picture.

For me to do a show that’s 110 dB for the front row, my PA has to produce about 119 dB at 1 meter. That’s right about what I would expect my compact setup to be able to do, with a small sliver of headroom. At 50 meters, my show has decayed to a still audible (but not “rock show loud”) 86 dB SPL.

That’s what I can do, and I’ve decided to be happy with it – because the folks I work with are likely to be just as happy with that as I am. People don’t hire me to cover stadiums or have chest-collapsing bass. They hire me because they know I’ll do everything in my power to get a balanced mix at “just enough” volume.

The Specifics Of The Show

Ultimately, the real brass tacks are to be found in what the show actually needed.

The show did not need 110 dB SPL anywhere. It needed a PA that sounded decent at a moderate volume.

The genre was folksy, indie material. A 110 dB level would have been thoroughly inappropriate overkill. At FOH control, the show was about 80 – 90 dB, and that was plenty. There were a few times where I was concerned that I might have been a touch too loud for what was going on. In that sense, I had far more than enough PA for raw output. I could have run a single pair of boxes and been just fine, but I didn’t want to get all the speakers out of the van and not use them. As I said before, I chose “double hung” to use all my boxes, and to use them in the way that would be nicest for people’s ears.


If you’re curious about running a double hung setup, I do encourage you to experiment with it. Curiosity is what keeps this industry moving. At the same time, you shouldn’t expect it to completely knock you off your feet. If you have a good-sounding system that runs everything through one pair of mains, adding another pair just to split out some sources is unlikely to cause a cloud-parting, ligh-ray-beaming experience of religious proportions. Somewhat like aux-fed subwoofers, going double hung is a taste-dependent route to accomplishing reinforcement for a live event. For me, it solves a particular problem that is mostly logistical in nature, and it sounds decent doing it.