Tag Archives: System Building

A Plan For Delays

I think this should probably work. Maybe.

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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Last year, I did a show at Gallivan Plaza that really ought to have had delays, but didn’t. As a result, the folks sitting on the upper tiers of lawn didn’t get quite as much volume as they would have liked. This year, I intend to try to fix that problem. Of course, deploying delays is NOT as simple as saying “we’ll just deploy delays.” There’s a bit of doing involved, and I figured I would set out my mental process here, before actually having a go.

Then, after all is said and done, we can review. Exciting, no?

So, here’s the idea:

A) Set primary FOH as a “double-hung” system. Cluster the subs down center, prep to put vocals through the inner pair of full-range boxes, and prep to send everything else to the outer pair. Drive the main PA with L/R output.

B) Have the FOH tent sit on the concrete pad about 60 feet from the stage.

C) At roughly an 80 foot distance, place the delays. The PA SPL in full-space at that point is expected to be down about 28 dB from the close-range (3 feet/ 1 meter) SPL.

D) Place a mic directly in front of one side of the main PA, and another mic in the center of the audience space, at the 80-foot line. (The propagation time to the delays will be slightly different depending on where people sit, so a center position should be a decent compromise.) Using both mics, record an impulse being reproduced only by the main PA. Analyze the recording to find the delay between the mics.

E) Send L/R to Matrix 1, assign Matrix 1 to an output, then apply the measured delay to that output. Connect the output to the delays. Also, consider blending the subwoofer feed into Matrix 1 if necessary.

F) Set an initial drive level to the delays so that their SPL level is +6 dB when compared to the output of the main PA. The added volume should help mask phase errors with the delays for listeners in front of the delay speakers, due to the contribution from the main PA being of much reduced significance…but it may also be possible that the added volume will be a problem for people sitting between the delays and the main PA. “Seasoning to taste” will be necessary. (For people sitting between the main PA and the delays, the time correction actually makes the delays seem to be MORE out of alignment than less, so the delays being more audible is a problem.)

So, there you go! I’ll let everybody know how this works. Or how it doesn’t.


Tuning A VerTec System

You can do a lot by simply treating it like everything else you’ve worked on.

Please Remember:

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

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I’m sorry that I haven’t been around much lately – I’ve been busy. Very busy. So busy that I’ve been saying “No” to things a lot.

One element of my busyness has been being turned loose on a VERY classy room in Park City. For the moment, I won’t name it here, although you may have heard of it. (Not naming it here might be a little ridiculous, actually. Anybody can get on my Facebook page and see what I’m doing. Well, anyway…) It’s a little too big to classify as a small venue in my own personal taxonomy, but hey, as we’re all learning, many of the lessons in this business scale up and down.

A task I was allowed to undertake was re-tuning the installed VerTec system. Some big complaints about it were an overabundance of “honk” and “boom,” and the hope was that I could do something to alleviate those problems. I believe I have mostly succeeded in making the rig better, and it was most definitely not an exotic process. I slapped a measurement mic in front of the FOH mix position, ran Room EQ Wizard, and got to work. The measurement traces confirmed what could be heard: The system was very heavy on the midrange, with some troublesome peaks in the subwoofer zone. After a bit of doing, we are where we are now, which is a much flatter place.

The main key, I can say, was to get over my own intimidation. VerTec, or really any similar system, looks hairy because of all the boxes involved. The thing to remember, though, is that for any given coverage zone the boxes are meant to combine into one big source. If you’re going to fret over something, fret over each overall zone of coverage, not the individual array elements. Pick your battles. As Bob McCarthy might say, decide what to tune for and ignore the rest. In my case, I had it pretty easy, because I chose to tune for the main room and not worry specifically about the boxes angled to hit people standing near the hangs. I didn’t have any outfills, infills, or other such coverage areas to consider.

A barrier that I encountered was that we’re locked out of part of the system management processor. With that being the case, I didn’t have the ability to adjust individual bandpass input or output levels. I did have EQ access, though, so that’s what I did all my work with. Was that an ideal situation? No, but what I’ve discovered over the years is that getting the basic magnitude response of a system to behave is the primary battle. I’m not saying other things don’t matter here. I’m not saying that adjusting bandpass gain by way of an EQ isn’t a kludge. I’m not recommending that, but I am saying that you might have to do it sometime, and it won’t ruin your life. Do what you can with the tools you have.

In the end, even with an imperfect approach, the system’s listenability has improved. We seem to be getting compliments on the sound in the room at a regular pace now. I’m certainly looking forward to next spring, when I plan to do another tuning that will start with tweaking amplifier gains first, but for now we seem to be in business.


The Pro-Audio Guide For People Who Know Nothing About Pro-Audio, Part 7

Amplifiers and loudspeakers bring us to the end of my series 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.

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“Now that we’ve turned audio into electricity and back again, we’ve reached the end of this series.”


This article is available, for free, right here.


The Pro-Audio Guide For People Who Know Nothing About Pro-Audio, Part 6

I believe in life after the console.

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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“The point of loudspeaker management is to make final, overall adjustments to console output so that devices which actually create acoustical output (speakers, that is) can be used most effectively.”


The rest of this article is available – free! – right here.


The Pro-Audio Guide For People Who Know Nothing About Pro-Audio, Part 2

The series continues with a discussion on cable.

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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From the article:

“The simplest and most robust connection possible is a single cable carrying analog electrical signals. Analog cabling is subject to many problems, of course, including noise induced by electromagnetic interference. However, its simplicity reduces the number of ways that an outright failure can occur, and the connection tends to degrade “gracefully.””


Read the whole thing for free by visiting Schwilly Family Musicians.


Is The Crossover Leaky?

A lot of low-end can still get into your mains.

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.

Every so often, I get to chew on a question that a reader asks me directly. I kinda wish that would happen more often (hint, hint, hint…). Anyway, I was sent a message on The Small Venue Survivalist’s Facebook page, asking if I could render an opinion on why a bass guitar seemed to have a surprising amount of LF information in the main speakers. The mains were being used alongside a subwoofer, with the sub providing a crossover filter at 100 Hz. What could the issue be?

There are a few explanations that would seem reasonable, if one discounts “catastrophic” issues like the crossover filter simply failing to operate as advertised.

1. Crossover filters, especially those implemented in active electronics, have a tendency towards a relatively steep slope. Even so, they usually aren’t brick-wall implementations. Everything below the cutoff doesn’t simply disappear – rather, it’s attenuated at a certain rate. With a filter set to roll off everything “below 100 Hz,” the mid-highs are still being asked to do a fair bit of work at the crossover frequency. The general vicinity of 100 Hz is actually quite bassy (depending on who you ask, of course), so the mains might be perceived as doing more than they should when everything is quite normal.

2. If a sizeable pile of low-frequency energy has been dialed into the bass-guitar channel, or the bass-guitar’s pre-console tone, that big hill-o-bass won’t be tamped down by the crossover. It will be split up proportionately, but following on from the first point, the mid-highs will still be tasked with reproducing their allotted piece of that big LF mound. Consequently, a surprising amount of energy may be present in the tops.

3. I have a suspicion that plenty of modern, two-way boxes receive some degree of “hyping” of their low-end at the factory. This makes them sound more impressive, and the manufacturer can get away with it because of safety limiters placed post-EQ. (The limiter prevents the low-frequency amplifier from supplying more voltage than the woofer can handle, and there may even be a level-dependent high-pass filter in play.) A low-frequency boost that occurs after the crossover reduces the crossover’s apparent effectiveness. Sure, the signal leaving the crossover might be down 12 dB at 75 Hz, but a +6 dB shelving filter put in place by the manufacturer at 100 Hz “undoes” that filtering to only lose 6 dB. Once again, a potential situation develops where the mid-highs are being asked to reproduce more “boom” than you expected.

It is entirely possible that an apparent problem isn’t covered by the three possibilities above, but they should catch quite a few scenarios where everything is hooked up properly and configured correctly.


Retort Report

Responses, and responses to those responses.

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.

Right after I posted my last article, somebody (that somebody being Jason Knoell of H2Audio) decided to REALLY kick the hornets’ nest and share the link in Stagehand Humor. Of course, I could not help myself: I had to read through the commentary and experience the reactions. The general themes, as well as some particular thoughts make for some excellent, extended discussion of the “Analog VS Digital” topic.

Please note that I’m not identifying individual users here. I don’t feel that it’s necessary, as this site’s main purpose isn’t to host community discussion anyway. (Such discussion is welcome and encouraged, but I don’t host the software to run it.) If you really want to know who said what, the discussion thread is here.

Also, I am 100% aware that I’m not going to change anyone’s opinion. That doesn’t bother me in the least. I’m writing this out so that people who haven’t yet formed their opinion can examine some viewpoints and decide whether or not they think I have a valid take on things.

I will try not to be too snarky, but I can’t make iron-clad promises.

Here we go.


“Have fun when your digital board crashes its operating system and you don’t have the flash drive on the truck.”

People say things like this as though analog consoles have never failed at terrible moments. Big, heavy, hot, expensive, rackmounted power supplies have been known to quit – sometimes spectacularly. Ribbon cables can get unseated. Channel modules can fail with very loud, showstopping results.

Hardware isn’t eternal, no matter its design principle. So…have fun when your analog console has the equivalent of a crash and you can’t fix it with anything as simple and convenient as a flash drive, plus you mightn’t have any spares (especially because analog is big, heavy, and expensive). Me? I effectively have two spare consoles that I carry with me, and the rig still costs less and weighs less than an analog counterpart.


“Say what you will but I can punch, spit, spill a drink, and blow smoke for 16 hours a day on a Mackie 1604 and still have a reliable board!”

An analog console MAY have an advantage in that damage to one section of the unit may not prevent the rest of the device from working. There may also be a period of time where component degradation due to repeated abuse isn’t immediately audible. “Integrated” digital setups tend to either work 100% or cease operating. That difference in failure behavior isn’t enough for me to take a technological leap backwards, however.

And being reasonably nice to a piece of equipment (rather than abusing it) is not as hard as some people might think. Be nice to your digital gear and it will last, as long as there aren’t any manufacturing flaws. Analog follows exactly the same rules, by the way.


“Forgot to address harmonic distortion, saturation of tone, or any acknowledgement of different consoles and their own signature coloration of tonality.”

I didn’t, actually, but I also didn’t go into much detail. I didn’t dig deep because I see running after that kind of thing as a giant waste of limited money and time. I’m not saying that it isn’t nifty when it’s there, if it’s working in your favor. The problem that I have is that the necessary premium to get it is vastly out of proportion to its utility. A console that’s just automatically magic with bass guitars and snare drums is a cool thing. A console that forgives being run hot, potentially in a way that’s even helpful at times, is also pretty rad.

I still maintain, though, that “baked in” signal coloration is basically a design limitation that happens to be fun. I personally prefer a console designed to be flat and clinical, where coloration of all kinds (possibly including distortion, if you’re into that) can be added with explicit intention by the individual operator. If I find I’m missing some sort of magical boost in the low mids – something that very rarely happens, but even so – I can always dial it up with the parametric across my main outputs.

And know EXACTLY what just happened.

You may not prefer that. You may be able to bear the direct and indirect premiums necessary to have an analog signal path that imparts a desired flavor to inputs automatically. That’s great! Don’t let me or anyone else get you down. All I’m saying is that gear which fits your workflow and not mine does not necessarily represent an inherent improvement in technology.


“Guy’s never heard an early 70s concert in a real theater with a real band.”

While I’ve never heard an early 70’s concert in a real theater, I HAVE heard real bands both in and out of various venues that I also consider pretty real.

And I wonder if it’s just possible that the sound of a great band, in a beautiful acoustical environment, playing to an appreciative audience, might represent sonic and experiential factors that are orders upon orders of magnitude more important than any inherent tonality imparted (or not) by the mix rig?


“Bad thing is you spent 20 grand on a console and by the time you were done figuring out all the routing and fx it was obsolete. That’s my biggest problem with digital.”

No, digital consoles are not obsolete the minute you get them. A new model may be waiting in the wings because development cycles are so fast anymore, but it’s not like the console that just got delivered won’t mix bands anymore, or is fatally flawed.

If you want to talk about a long-term ecosystem of support, spares, rental-stock, and add-ons, I can see where you’re coming from – but in all cases, that kind of thing only comes about for the mix rigs that have gotten picked as favorites by the industry at large. Consoles are like pop-stars and rock bands. We remember those that stood the test of time, and conveniently forget that lots of analog and digital offerings didn’t manage to spark, and thus never generated that kind of sustaining ecosystem.

By way of example, I have a pair of Tascam DM24 consoles that sound just fine, and work just fine. I mixed on them for years. They never had the following that the Yamaha 01V series had, though, so I was basically on my own in terms of support and ancillaries. They were “obsolete” even when I got them, in the sense that Yamaha had handily passed them by. So what? They were still powerful tools.


“When analogue peaks, you get “warmth” or natural distortion. When digital peaks, you get clipping and digital breakup.”

Both events being described are an overload. Both are distortion/ clipping. The phenomena on display are not fundamentally different, though the specific tonalities of the events do differ.

My question is: Driving your console’s main bus into clipping isn’t a best practice. Why are you doing it so much that the console’s ability to forgive your gain structure is a main factor in your purchasing decisions?


“Small venues can’t afford digital that doesn’t have latency issues.”

I once built a digital mix system that had a roundtrip latency of about 9 ms, as I recall. That’s really not the best situation…but I used that system for years at Fats, and nobody every complained about it. Mostly, they raved about how great the shows there sounded, both on and off the deck.

My new, non-homebrew rig has a stated latency of about 1 ms. Nobody’s complaining about that either. Latency is a convenient audio boogeyman that gets blamed for all kinds of problems that seem vague or unexplained. It really isn’t as huge a factor as it’s made out to be, and it certainly does not account for all the ills that some folks love to attribute to digital.


“We had a mid level digital board that when pushed just broke up and sounded terrible. We had to boost the processing on the amplifiers and run the mixer as low as possible.
It sounds like the guy writing this article is new to sound and maybe has not used pro equipment!”

If you’re “pushing” any console, analog or digital, in order to drive the PA to full power, your system gain structure is wrong. It’s especially wrong if you’re pushing the console into clip. On the dBFS scale, the region around -20 is the equivalent of “nominal” level on an analog console.

Digital systems, as a rule, do sound horrible when clipped. So, don’t clip them. It’s really not hard to get yourself into the mindset.

I know this stuff because I’m NOT new to audio. I know this stuff because I’ve had hands-on time with consoles that cost everywhere from $50 – tens upon tens of thousands when they were new. I’ve never been bothered by the sound of any of them. I’ve never had a religious experience because of the sound of any of them either. It’s because I’ve used real equipment on real shows (gigs that play to a couple-hundred patrons are VERY real, by the way), and have had to make real purchasing decisions with real money – that is, my own money – that I’ve come to my conclusions.

A case in point is a story that I’ve told several times, in several forms. My schooling was when I had my major, hands-on experience with spendy, large-frame analog desks. Next to the big, premiere, “A Room” was the new “D Room” with a pair of Tascam digital consoles. The two Tascams together were about $6000. The A Room SSL was the high-rent behemoth. Material in both rooms sounded plenty nice. The consoles in the D Room, though, did nearly everything that the SSL could do. They did it more easily, and faster, and for maybe 10% of the cost (if the percentage was even that high).

In real life, convenience, features, and affordability are vastly more important to a console than “It seems to sound super nice under certain circumstances which may not really be the direct result of its technology base.”


“I could make a very good mix on the analogue board in a large venue where not as much fine tuning and adjustment is necessary as you’re not battling stage spill and close proximity to PA as much…The analogue does sound warmer, the mix sounded slightly fuller and for just an acoustic act or a simple band I can still make a mix sound awesome!

Once again – the factors being described here as making a huge difference have nothing to do with the console’s technology base. They are environmental and circumstantial, which hold vastly more sway over the sound of the show.

Also: If you didn’t do exactly the same mix, at exactly the same SPL, of exactly the same band, in exactly the same room, with exactly the same audience, you can’t seriously claim that the analog console was the primary reason it sounded “better.”


“Simple fact– you cannot digitize EVERY bit of sound.”

Simple fact: Yes, you can, and we have been for a long time. Even 44.1 kHz systems reliably capture the entire audible spectrum, and 24-bit converters have dynamic range that (to my knowledge) continues to outperform the analog input stages they are necessarily mated to.


“But it’s better to teach in analog, it requires you actually to listen not just look at a screen.”

Consoles had labeling and useful meters, and signal analysis devices did exist before digital audio was “a thing.” Besides, “listening only” can trick you. Combining your ears and your eyes – and making sure they agree – is a powerful tool for doing better work as an audio human.


“Sacrificing quality for convenience is all it is. If analog is so bad, why are there so many Midas consoles on the road still? Hell, Bonnie Raitt was out with a Gamble console. She sounded better then any digital board could get her to sound.”

No, actually, it’s choosing to have quality, convenience, and features at a great price-point over getting quirks at high expense.

And I’m not saying that analog is bad. I’m saying that it’s not better. There’s a difference.

There are a billion Midas desks on the road because they ARE good consoles. They are consoles that people know how to use, and associate with good sound. I never said they weren’t. I’m also saying that the cost of obtaining, maintaining, and transporting them is hard to justify – unless you’re a touring company or rental house, of course, and everybody who calls you wants one.

And I will close by saying that Bonnie Raitt is a master performer with a killer group of musicians at her side. That matters far more than the console ever could, and I don’t see any way to practically back up the (“handwaved” at best) assertion that her performances would sound any less than brilliant through a digital desk.


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.

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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.