Tag Archives: Loudspeakers

An Adventure In A Peavey

Sometimes, a broken thing is less broken than you might think.

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Remember that road gig I was talking about a few weeks ago? Well, during our setup for the first day, I noticed something. One of my monitors had a cooling fan that wasn’t being much of a fan. It was just sitting there, not moving any air molecules.

This was disconcerting to me, but there wasn’t much I could do about it. We ran the show with some signal going through it, and it apparently did not “thermal,” so – good on ya, PVxP!

Of course, even with the box having behaved itself, I couldn’t just leave it be. I wanted to know what was wrong, and if it could be fixed. So, I gathered up a few tools and went on a little adventure.

This monitor had spent its previous life at Fats Grill. During that life, the fan had ingested a fair amount of “old Salt Lake City basement gunk,” which is a particular kind of evil dust that’s nearly impossible to clean up, and adept at invading everything. I imagine that every city has its own version of this stuff. At the outset, I was pretty sure that the poor fan had gotten all bound up with this goop, and might have burned itself out.

But, to be sure, I would have to get the back panel off the speaker and do some rooting around.

IMPORTANT: Poking, prodding, digging, tweaking, yanking, or otherwise messing with the internals of gear that runs on “wall” power can injure or kill you. I am NOT responsible if you attempt activities like this and end up surprised, hurt, or dead.

The amplifier module connects to the rest of the speaker through this little bit of fun. The four-conductor part seems to be what mates with the drivers (the clues being labeling like HPF + and HPF -). As far as I can tell, the little two-wire connector is to do nothing more than light up the LED in the front of the box.

Peavey just couldn’t resist gluing that LED connection shut. Getting it out meant pulling the entire contact assembly off the circuit-board pins. Geeze…

An interesting design choice with these Peavey PVxP speakers is the “big tub of nothing” that the amplifier module screws into. All that’s in there is the little tiny PCB that acts as the bridge between the backplate electronics and the rest of the loudspeaker. I can guess that it makes sense from a modularity standpoint, as all the negative space seems to be able to accommodate either an active electronics unit or a passive crossover setup with no fuss.

Nobody in here but us drivers!

And some wire, and an LED, of course.

After extricating the amplifier module fully from the enclosure, I was able to get a look at where the fan got power. My first step in troubleshooting was to get the fan off the main chassis, in the hopes that I could figure out why it was no longer inclined to spin.

Much to my surprise, applying power to the amp module caused the fan to run like nothing was wrong at all. The hub was turning as smooth as glass, with no noise of rubbing or anything else being amiss. Well, that was surprising – but in a nice way. I had envisioned having to find a replacement fan, and maybe do some wire splicing to get back to full operation. It didn’t seem that I would have to do any of that now.

Since I had gone to the trouble of getting the major assemblies apart, I did want to satisfy my curiosity as to how much voltage was used to drive the fan. (If the fan would have refused to turn at all, my next step would have been to determine if it was getting any power.) Using my meter to take a reading across the pins got me about 33 VDC. Apparently, Peavey runs the fan a bit “hot,” because…

…the actual fan is a 24 VDC model. (Then again, the power supply tap could be a little “off.”)

Side note: When you shut down the amplifier module, these LEDs stay lit for a while. I imagine that could be code for, “The big capacitors on this thing still have PLENTY of charge in ’em, pal, so don’t touch anything right now!”

I screwed the fan back onto the amp housing, and THAT’S where things got interesting. I applied power, and…nothing. The fan was at a dead stop again. I wondered if there was something about the fan’s orientation that was giving it trouble. I got out my screwdriver, and started loosening the fan from the mounting holes. Suddenly, with a bit of a grating scrape, the fan sprung to life again! I ran a couple of the screws back in a turn, and the blades ground to a halt.

The problem the whole time was that the chassis had been pressed too tightly against the fan hub. As I said before, the apparent behavior of PVxP fans is to pull air into the enclosure. In the case of these fans, that means that their integrated “cage” faces in, instead of out. As such, the outer plate can pretty easily be brought into contact with the unprotected side of the fan hub, and that can stop things pretty efficiently.

I backed a couple of screws out just a touch, and what do you know – I had a working fan again.

With the amplifier re-mounted to the box, I ran some music through the enclosure. Everything seemed fine, and that made me a happy audio-human.

Double Hung Discussion

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

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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:

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.

Thermodynamics, System Coverage, And The Cost Of Lunch

Lunch is not free, and energy isn’t magic.

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Before diving into this topic, I want to be very clear on a few points. First, this kind of discussion is a bit “above the pay grade” of small-venue folks like myself. Second, there’s a lot of theory involved, because I don’t have anything in the way of deep, direct, hands-on experience with it.

Ready a grain of salt to take with all this, okay?

Okay.

The pro-audio world sometimes likes to behave as though thermodynamics is less of a harsh mistress than it really is. That is, there seems to be a semi-willful ignorance regarding energy and where it goes. This can lead to a sense of there being some sort of free (or reduced cost) “lunch” when it comes to the directivity of a system. The problem is that lunch is always served at full price. If you want sound to only go where you want it to go, you also have to deal with the laws governing the behavior of that audible energy.

Achieving useful, desirable directivity with an audio system was traditionally the purview of wave-guidance. In other words, horns. You channel your sonic flux through the horn, and (within the physical limits of the horn), you get certain advantages. One benefit is better “pressure transfer” to the world beyond the driver. Another nice bit of help is greater directivity. With a horn of the correct overall size and flare rate, you can focus sonic energy (within a certain passband) into a defined radiation pattern.

When horns and horn-cone hybrid boxes are used with the intention that their natural, physical directivity prevents them from interacting too much with each other, what you have is a point-source system. In such a setup, the hope is that any particular listener is overwhelmingly hearing only one source per passband…or, even better, hearing all passbands from one source. (This only has so much feasibility, especially where low-frequency material is concerned.)

As the ability to use more boxes and more electronic transformation has expanded, people are doing more and more with system processing on arrays. The enclosures involved in these arrays also have natural, physical directivity. They are also very likely to use some sort of horn for the high-frequency section. Unlike a point-source system, though, the idea is that you actually are supposed to hear the boxes interacting. This interaction can be controlled on the fly by way of changing box or driver amplitude and delay. If you want one kind of coverage, you tweak the system to interact in one way. If you suddenly decide that you want different coverage, it’s theoretically possible to simply tweak some parameters and get your change.

This is very nifty. Managing everything with actual, physical horns is a heavy, large, and predetermined sort of affair. Processing changes, in contrast, are flexible and physically lightweight. (The math, on the other hand…) “Nifty” is not “magic,” however, and this is where some people get tripped up.

The Lighting Analogy

Bear with me for a moment, as we do a foundational thought experiment.

Let’s say you have a stage light. You turn it on, and it works nicely, but you have light energy hitting something you don’t want to hit. The nice thing about your fixture is that it has shutters. You adjust the shutters so that the light no longer falls on the undesired area.

Question: Did the light falling on what you actually wanted to hit become more intense as you shuttered the beam?

No, of course not.

The visible-light radiation from the fixture hit the shutters, and was largely exchanged into heat. The luminous flux wasn’t redirected through the business-end of the fixture and mystically redirected – it was absorbed and converted. The relevant thermodynamics of the system are fully in play, and inescapable. The “cropped” energy was simply prevented from reaching a target, and that energy stopped being useful as visible light.

Now, let’s take a different approach. Let’s say you could avoid hitting an unwanted area with the light by a different means: Optics. You put a lens with tighter focus into the system, and restrict the beam-width that way.

Did the light falling on the object become more intense?

Yes, all else being equal.

The lens took the entire output of the fixture and focused that flux into a smaller area. The maximum possible fixture output remained usable.

So, what does this have to do with sound?

Focus Vs. Cancellations

In an effective sense, a horn is acoustical “lensing.” It’s a way to focus sonic energy from a driver (or drivers) into a defined space, physically giving you the directivity you want.

The flipside to this is a large, highly processed array of sound sources. Given enough drivers, enough processing, and enough time, it seems entirely feasible that a system operator could get the same coverage pattern as what would be found with point-source boxes. What has to be remembered, though, is that “lunch” has a required cost. The thermodynamics of the two approaches are not the same at all. Like our hypothetical light and tight-beam, hypothetical lens, the highly focused horn is energy efficient. A single driver (or set of drivers) have as much of their acoustical output as possible put to use solely for covering an audience.

The big, technically advanced array is energy inefficient, because it doesn’t use a physical object to focus its coverage. Instead, it requires the interaction of more energy. If you want to create an acoustical pattern through interference, you have to combine the output energies of multiple audio-output units. There are many shades of grey to take into account, of course. Even so, in the most extreme case, cancelling the output of a 1000 watt driver may require the use of another 1000 watt driver. The energy consumption of the resulting system is 2000 watts plus inefficiency losses, but your usable sonic output has not necessary doubled – remember, you’re using one driver to cancel the other for purposes of pattern control. At the physical point of that cancellation, the usable sonic energy is 0, even though the system is still consuming a large amount of electricity. It’s the same as shuttering the light. The sonic energy is merely being made unusable in a certain target area.

…and there’s a tendency to try to forget or “talk around” this. Marketing departments especially love to come up with fancy terms for things, even when those terms make no sense. Some of these highly processed systems are called impressive things like “complex point source.” The problem is that there’s no such thing. As soon as the idea for the system is to have large, intentional, audible interaction and interference across multiple units producing wideband audio, we aren’t in point-source-Kansas anymore, Toto.

There’s nothing wrong with that. Systems that have their coverage managed by way of processing and multi-box interactions are a great tool for versatility. You always bring the same gear and deploy it in basically the same way. Having exactly the right boxes for a needed point-source solution is much more possible when you’re doing a permanent, custom-built install. I’m inclined to believe the folks who claim that point-source will always measure as being more clean and coherent, but I also believe that measuring well isn’t the end-all, be-all in a discipline that has so many trade-offs.

The solutions are different, their appropriateness is situationally dependent, they are not thermodynamically equivalent, and someone is going to have to buy lunch.

All Powered Speakers Are Not Created Equal

March’s Schwilly guest post.

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“One pitfall, though, is that the label of “powered” on a box is what I call a “sloppy metric.” Because a good number of active speakers truly are packets of highly engineered, carefully tuned technology, it becomes easy to assume that all specimens able to be referred to as “powered” share similar traits.

This is not the case.”

The entire article is available (free!) at Schwilly Family Musicians.

For The Love Of Trim Height

Trim height is very helpful if you can get it (and do it safely).

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There are a couple of things I have to say before I dive into this:

1) I’m only going to get into the trim-height factors that, in my view, have a “snowball’s chance” of being helpful in a small-venue setting. Trim heights above 30 ft, with multiple, arrayable boxes in use can be a very handy thing…but they also involve venue size and PA deployment complexity that’s beyond the scope of this site.

2) The higher the trim height, the more dangerous things can be. I don’t say this to frighten anyone, but rather to call to mind the safety issues. Setting a box on a solid deck is very safe. Putting a box on a stand is less safe. Suspending a box is even less safe than putting the box on a stand. Higher trim heights, done with proper attention to weight limits, stability, and appropriate equipment, can be done “safely enough.” Not paying attention to those factors can result in someone being seriously injured or killed. Under NO circumstances should you compromise safety for an acoustical outcome. (I accept ZERO responsibility for anything you try, just to be clear.)

3) I’m using simplified drawings to make my points. They don’t exactly represent what sound does with real loudspeakers, real people, and a real room. However, the approximations are close enough to talk intelligently about what’s going on. (Just for a start, a box that claims 40-degree vertical coverage actually has a great deal of output beyond 40 degrees – as you have no doubt observed in real life.)

Anyway…

Trim height – that is, “gettin’ speakers in the air” – has real advantages. Done correctly, it lets you maximize the use of your loudspeaker’s output, minimize the amount that the PA “excites” the room’s acoustics, and use the acoustical impedance of your audience to an advantage.

Low Trim

We’ve all seen this at some point. An audio rig gets set up so that it’s sitting directly on the stage. It’s easy, cheap, and very safe. The loudspeakers are highly stable, and if one does get knocked over, it will probably hit someone’s foot or leg at low speed.

There are real problems, though. The biggest one is that the acoustical impedance of the audience is working against you. Electrical impedance is opposition to current flow. Acoustical impedance is the opposition to sound-pressure flow. Humans are pretty decent at absorbing sound, which means that firing a speaker directly into the front row is a waste of power. For all intents and purposes, the humans in the way of that audio are acoustical resistors, all lined up in series. A sonic “shadow” is cast by the people blocking the direct path of the loudspeaker’s output.

The upshot is that you can use up a ton of available output on trying to “push across” that absorption. Also, the front row gets a very different show than the listeners at the back. The folks in front are getting an experience with lots of direct sound, whereas pretty much everyone else is getting very different volume and a high proportion of indirect sound. The fictional venue I’ve constructed has a 20 ft ceiling, but it’s easy to imagine one with a much lower roof. Cut the ceiling height in half, and the direct sound that doesn’t hit the front row just hits the ceiling and starts bouncing around.

The thing is, we want to use our output to hit listeners, not boundaries.

Speakers On Sticks

The next step is to do what’s practical for most of us: We put boxes on tripods.

This takes a little bit of doing, costs extra, and also requires some thought to safety. If a tripod falls over, someone could get hit in the head with a heavy piece of equipment; Due diligence is required.

Even so…immediately, you can see that the consistency of experience from audience-member to audience-member is greatly improved. Yes, the people in front do still generate acoustical shadowing, but the obstruction is far less pronounced. Pretty much everybody has a good chance at hearing the direct sound from the loudspeakers.

There is an acoustical downside, though. Getting the speakers in the air has increased the amount of output which is hitting the room’s boundaries. The reverberation we’ve introduced into our mix is rather greater, and we’re also firing output into a lot of nothing (before the output arrives at a wall or the ceiling, of course). If the ceiling is low, a lot of the loudspeakers’ energy is splattering against it. The situation is a waste of power, but at least it’s not as big a waste as trying to “blow through” the front row of spectators.

Just Hanging Around

What if we could get our boxes about 12 ft in the air, and angle them downwards?

This is spendy and risky. You’ve got to have the proper rigging hardware, and whatever you rig to must be durable enough to handle the load. If the suspension system fails, a very heavy object could be moving very fast, and on a path towards somebody’s skull. The consequences for getting this wrong are high, so it shouldn’t be attempted without careful thought and professional help.

If the logistics are handled properly, then major advantages are conferred. Pretty much all of your output is being directed towards actual people. The audience obstruction of direct sound has been further reduced, meaning that there’s an even higher chance for everybody to be getting the same show. Our output is largely directed away from room boundaries, which means less indirect audio to reduce mix intelligibility.

This is also the configuration where the audience’s acoustical impedance works in our favor the most. A lot of the room reflections are likely to encounter a human’s absorption at an earlier time, further reducing reverberation intensity and the accompanying loss of intelligibility. Using our audience to soak up what we DON’T want, while letting them listen to what we want them to hear is a win-win.

A box that’s safely in the air and pointed in the right direction does more work that’s actually effective.

You Can’t Just Pile It Up

More speakers will get louder, but they won’t necessarily sound better.

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I once helped out with a children’s concert.

Don’t laugh – it was pretty cool. The performer was a real pro, and very gracious. There were plenty of (knowledgeable!) volunteers around to lend a hand, and the space we were in was pretty darn classy.

The PA was, surprisingly, not so nice. That it was not so nice was a bit weird, because the loudspeakers deployed were significantly more spendy than what I would personally have on hand. The thing was, though, that no matter what I did at the console, the sound of the show just wouldn’t clear up. The vocals, in particular, vexed me. They didn’t sound awful, but they weren’t nearly as intelligible as I would have liked.

(The thing with shows for kids is that the lyrics are even MORE really important than at other kinds of productions.)

The loudspeakers in the room were deployed such that an “inner” pair covered the central seating, and an outer pair covered some additional audience wings to the sides. At one point, I discovered that I could mute the outer pair.

Instantly, the show sounded a lot better. It was not subtle at all. The un-tameable mud basically went away.

And then, with a mental sigh, I unmuted the outfills. I knew that it was inevitable that someone would sit there, and they would want to hear some coverage from those boxes. There are many situations where “coverage” gets just as many votes as “quality,” and this was one of them. I couldn’t ask the organizers to rope off those sections, especially as I had no idea of the expected attendance, and I was also going to have to hand off the console to the house crew for the actual show. As I said, the volunteer team was knowledgeable, but I’ll also add that there were some communication barriers…not the least of which being that we had “a lot of cooks” in the kitchen. I couldn’t just make a decision, say, “this is how it’s going to be,” and expect everybody else to do my bidding.

Anyway…

The real problem was that the owners of the room had “piled up” (or had let someone else pile up) a bunch of PA in order to solve a coverage problem. To be fair, it could have been a LOT worse. The outfills were at least decently splayed away from the inner pair of boxes, and that helped, but the harsh reality remained: The system sounded better overall when half of it was muted.

More PA is not necessarily better. More PA will get louder, and sometimes you need that. More PA can cover a wider area, and sometimes you need that. At the same time, though, more PA can cause you some real sonic problems.

Tickets To The Splatterfest

You may remember me talking about how I dislike “AV” systems that just throw a bunch of sound everywhere. You may also remember an article I wrote about how acoustics really do matter, and that EQ can’t actually fix acoustical problems.

One problem with more PA and more coverage is that, unless you’re very careful, you end up making your acoustical problems worse.

In live audio, one of our greatest enemies is indirect sound. Indirect sound is the collection of sonic events that have traveled to a listener AFTER hitting something else, like a wall, ceiling, or floor. Direct sound, on the other hand, goes straight from the loudspeaker to the observer.

At this point, you might be wondering what I’m upset about. By adding more PA, you’re hitting more audience members with direct sound, right? Well – think about it for a minute. Yes, you are getting more direct sound to a section of the audience, but in a significantly reverberant room, you are also producing more reflections that are very definitely audible to everybody else. For a few people, the solution may be a bit better, but to most of the audience the solution is actually worse. It’s even more garbled and smeared than it was before.

When you started, the average audience member might have been hearing direct sound and, maybe, five secondary arrivals. Now, they’re getting something like ten or fifteen secondaries (or whatever, I’m just guessing), which are starting to swamp the direct acoustical signal. Uh-oh…

A Dearth Of Directivity

Another bit of trouble that us small-venue types run into is that our loudspeakers aren’t really meant to “array.”

There are loudspeakers in this world that are meant to be “piled up.” They’re designed for it. They have high directivity, meaning that a lot of effort has been put into getting the box to radiate a great deal of output into certain horizontal and vertical angles, with a lot less output spilling outside of those areas.

To my knowledge, actual, perfect directivity is impossible. With lower frequencies, getting “perfect” directional control requires enormous, and enormously impractical physical size.

Even so, some boxes are far more directional than others. Most of the enclosures that folks like me encounter are very NOT directional. They’re made for a market that requires system simplicity and compactness, with wide coverage from a small number of boxes. These not-super-directional loudspeakers make it easy to hit a wide swath of listeners, but all the “spray” makes it hard to MISS things. Like room boundaries.

Also, remember what I said about lower frequencies requiring large physical size for pattern control. These little boxes that we use become more and more omnidirectional as the frequency of our program material drops, which means the LF “garble” is going everywhere. That low frequency information goes pinballing throughout the room, and it also finds a direct path to other listeners. That unintended direct information combines with the intended direct information of other boxes, and it also sums with all the indirect information, and all this can quickly turn into a giant ball of muddy woof and boom.

Let me show you:

Let’s say you have the most amazing, wide-coverage loudspeaker ever invented. On-axis, its transfer function is perfectly flat.

Sweet! But now, you put another of that box into operation, firing off to the side. The top end is rolled off, and the room acoustics build a peak at around 400 Hz.

Now you add another one. (By the way – what I’m showing you is likely to be far more tame than what you would actually get in real life.)

The weirdness buildup collects in a BIG hurry. If you’re getting the idea that this kind of thing would very quickly become tough to manage, even with a decent EQ, you’re quite right – and remember that EQ can’t fix your multiple arrivals problem.

Running Interference

Another wrinkle to get under your toenails is the problem of time. The direct sounds from the various boxes that are being fired into the room don’t all arrive at each listener at the same instant. If all those boxes are reproducing the same signal, the interference caused can be on the order of “astounding.”

For instance, let’s say you have a pair of those super-perfect loudspeakers. Stand between them so that the sounds from both boxes arrive at exactly the same time, and the boxes sum with beautiful coherence.

…and then, you move so that one box arrives 1 ms later than the other. The box you’re nearer to is a bit louder than the “late” enclosure, but that isn’t enough to fix what happens:

Then, you add a pair of outfills. One of them arrives 2 ms later than the reference box, is rolled off, and has the bump at 400 Hz.

The other one is 3 ms late.

Every box that gets added is making things worse, not better.

Now, what I don’t want to do is to sound an unneeded alarm. In the theoretical case, yes, the very best “in the room” PA system is a single, perfect box that can cover every audience member. The theoretically perfect case, of course, isn’t going to come along anytime soon. We use “spaced pairs” of boxes all the time for a very good reason: The extra overall coverage and power is what we need, and we can manage the imperfections in various ways. Also, it’s important to remember that multiple boxes can be used successfully as part of a holistic solution. They have to have the right directivity, have the right EQ applied, be pointed in the right direction, and be used in an acoustic space such that all factors together result in each audience member hearing predominantly one box over all the others. Big shows use multi-box setups that are planned and executed carefully. The systems are designed to array in one fashion or another – some even being crafted such that the box-to-box interference gets a desired result.

(Honest-to-goodness line-arrays, for instance, rely on box-to-box interactions to keep the overall vertical coverage narrowed into a desired area.)

The takeaway here is that just throwing more PA into a room without a lot of thought is going to cause problems. Yes, the whole thing might get impressively loud. Yes, you might get it so that everybody in the room is hearing a lot of SOMETHING. It might sound terrible, though, and at some point the tradeoff becomes unacceptable.

VRX Brackets

One word: No.

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Before I begin, I want to make it clear that I do not have any “hands-on” time with the VRX series by JBL. However, I know enough about how they are supposed to be used to “be dangerous.” Also, depending on your perspective, this may not really be small-venue material.

Now then.

It has come to my attention that some folks are using 3rd Party and/ or homebrew suspension brackets so as to defeat the built-in angle on VRX loudspeakers. That is, VRX loudspeakers naturally array in an arc, and there are people out there who are arraying them in a straight line.

Please, do NOT do this.

The first thing to talk about is the safety problem. I am not one to say that different and weird things can’t be tried if you’re careful. However, suspending loudspeakers anywhere that a rigging failure could cause injury or death is not a trivial matter. Such a situation is generally inappropriate as a test lab. Also, if something does go horribly wrong, using ONLY approved hardware is far less of a liability than deploying a non-manufacturer-approved solution.

If you are using rigging hardware that is not approved and endorsed by JBL for mounting VRX boxes, then stop.

The next thing to talk about is the audio side, and also the perception side.

A VRX system is a “constant-curvature array.” JBL even says so. JBL also calls VRX a “line-array.” However, everything I have read on this subject (mostly commentary from people who are far higher-up in this business than I am) indicates that the two terms are not actually compatible. A constant-curvature array is a vertically-oriented point-source deployment. It is not meant to behave as a classical “line source,” although the boxes will interact greatly at lower frequencies. I strongly believe that JBL labels the VRX system with the line-array name because of marketing: People associate “line-array” with “better” or “professional,” so there’s an incentive to refer to a vertically-deployed loudspeaker system as a line-array.

VRX hangs in an arc because it is supposed to. It is designed around that kind of deployment. Defeating the built-in angles and hanging the boxes straight down is against the entire design concept of the system. The boxes are not designed to array that way acoustically or physically. A straight-down hang of VRX causes the box outputs to interact (and interfere) in a way that is actually unhelpful in terms of total audio quality. It may be that a straight hang gets somewhat louder, but the phase interactions – especially at high frequencies – really aren’t what you want.

If an actual, JBL, multi-angle-capable line-array is what you want, then buy a Vertec system. (Or, if you want a system that only hangs straight-down and manages coverage through processing, look into Anya.)

Once again, please understand that I do encourage experimentation and “weirdness.” However, in the case of highly-engineered loudspeaker systems, I must very much recommend that they be treated like medication. (Use only as directed.)

The Lambda Effect

Wavelength is precipitated by frequency, instead of the other way around.

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After I published my article on big drivers and low frequencies, I got an email. In that email was a query about why bass drivers don’t have to move, say, 22.5 feet in order to produce a 50 Hz tone.

It’s a fair question, especially when you look at things like organ pipes. Making a low tone requires a big, long pipe, so why don’t bass drivers have to have a similarly long excursion – that is, forward-to-back movement measured in feet or meters?

We’re Trying To Do Different Things

The first part of the answer has to do with different technologies of sound production. In an organ or human-held wind instrument, we’re trying to produce a tone by setting up a vibration in an air column. The vibrating mass of air IS the “driver,” in a practical sense. With a conventional loudspeaker, the point is to produce a tone by vibrating a diapragm into free air. Although we may use all manner of horn-loading to create an acoustical impedance transformer, we’re not really trying to turn the loudspeaker into a flute or trumpet. We’re just trying to make the loudspeaker “play nice” with the surrounding atmosphere.

You might say that, in a wind instrument, air is the source AND the medium. With a speaker, air is really the medium only. A vibration is created and travels away from the source, and that’s the key to the second part of the answer.

Counting Cars

Let’s say that you have two parking garages full of cars. These parking garages are magic, because vehicles exiting them come out the exit ramp at a preset speed of 40 mph, and then hold that speed. Your task is to count how many cars come out of each parking garage every minute.

Parking garage #1 lets a car out every five seconds. Garage #2 lets a car out every 10 seconds. Garage #2 obviously has a lower frequency than #1: Over the course of a minute, #2 spits out 6 cars to #1’s 12.

Did you notice that measuring the distance between each car isn’t necessary for the “count how many cars exit each minute” objective? The frequency is what matters to you. The distance between cars is necessarily affected by the frequency that they exit the garages, but that’s not what you’re directly looking at. The slower garage has a larger distance between cars, of course, which is caused by it taking longer to disgorge an automobile. If all the vehicles are traveling at the same speed, then a lower exit frequency results in a longer space between each of those vehicles.

My example is definitely simplified. The speed of sound isn’t exactly the same everywhere, all the time. All kinds of interesting “lensing” phenomena can occur when sound waves travel across a temperature gradient and change velocity. At a coarse level, though, the principle is the same. Perceived pitch has to do with frequency, as opposed to wavelength. We don’t need a minimum amount of path length in order to hear a certain tone, which is why headphones can produce serious bottom end while being right up next to (or even inside) your ear canal. In any conventional, live-sound application I can think of, frequency precipitates wavelength instead of the opposite. A vibrating object creates pressure waves that travel outward at a certain speed, and so the disturbances caused by very fast vibrations don’t get very far before the next amplitude cycle begins.

(Even when encountering the Doppler effect, which is a special case of changing wavelength by moving the source or listener relative to each other, the thing that matters to our perceiving a drop in pitch is how often the positive and negative pressure fluctuations arrive.)

Wavelength matters a great deal to acoustics, but it doesn’t produce frequency as far as a driver sitting in a box concerns us. Rather, frequency produces wavelength. “F” is the cause, and “lambda” is the effect.

Why Do We Use Big Drivers For Low-Frequency Material?

It’s easy to say that we have to move more air, but there’s more to it.

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There’s a certain intuitiveness to the idea that a subwoofer driver (especially one that radiates directly, as opposed to being horn-loaded) is big. Or rather, that the subwoofer driver has a large diaphragm relative to a high-frequency driver. If you want a low-frequency noise, and you want it loud, it just makes sense that you need something big for making it. Bears, for instance, have lower voices than doormice.

For the average person in entertainment, I don’t really find satisfaction with the standard explanation for why we use big drivers to produce LF information. We say things like, “Ya gotta move more air, dude!” and then move on. Sure we have to move more air, but WHY? It doesn’t help that the topic seems to be avoided by many sites that talk about sound. My guess is that it probably has something to do with the physics being more hairy than a lot of audio humans are ready for. It’s the kind of material that you’d expect to find in acoustical engineering classes, as opposed to a live-sound engineering course. (Acoustical engineering is “classical” engineering, whereas being a sound engineer for entertainment emphasizes equipment operation.)

As such, even folks like me end up “feeling around in the dark” in regards to the question. We know that there’s more to it all, but how it works is tough to piece together.

This article is all about me trying to piece it together. A big “thank you” is due to Jerry McNutt, an honest-to-goodness Product Design Manager at Eminence Loudspeakers. Two years ago(!), he was kind enough to answer some of my questions about this topic, and I’ve been chewing on those answers sporadically since then.

Anyway…

Please be aware that this is a “best attempt.” My conclusions may not be exactly correct, but I don’t have an easy way to really verify them. Treat this all as food for thought seasoned with at least one grain of metaphorical salt.

Sound Intensity vs. Frequency

Intensity is a measure of power over area, or watts applied per square meter at the observation point. Most of us don’t think of sound level in terms of intensity as defined by physics. We’re used to dB SPL. Conversions are definitely possible, but that’s not the point here. The point is that intensity does relate to frequency, and greater intensity means that something is perceived as being louder.

If you want to actually calculate intensity of sound with real units, there’s a fair bit of math involved in figuring out how to do so. The end result of all that figuring still looks a bit intimidating to those of us used to moving no more than three terms around. According to the physics.info site:

I = 2π^2ρƒ^2v∆x^2max

But…if all that’s desired is to make comparisons regarding how intensity varies with frequency, everything that isn’t “ƒ” can be set to a value of 1:

I (abstract comparison) = ƒ^2

If we start with good ol’ 1 kHz as a reference point, the abstract comparison intensity is 1000^2, or 1,000,000. If we go down an octave, the frequency is 500 Hz. Five-hundred squared is 250,000.

In other words, if everything else but frequency is held constant, then going down an octave means the sound intensity drops by a factor of four.

To really drive this home, let’s consider the frequencies of 60 Hz and 6000 Hz. We would generally expect the low side to be produced by a big ol’ subwoofer, and the high side to be in compression-driver territory.

I (abstract comparison) = ƒ^2 = 6000^2 = 36,000,000

I (abstract comparison) = ƒ^2 = 60^2 = 3,600

36,000,000 / 3,600 = 10,000

In terms of power, a factor of 10,000:1 is jaw-dropping. Pushing an itty-bitty compression driver with one watt is common. Pushing one with 10,000 watts, well…

Two vs. Four

From the above, I think you can get an idea of the importance of “moving more air” to keep everything manageable. We have to do something to counteract the intensity drop from lower frequency. It’s actually a multi-factor problem, of course, because real-life tends to be that way. We can move more air by making a driver undergo longer excursion (forward/ back movement), but there’s only so much that’s doable. Closely related to that is more drive power. That’s good, but again, there’s only so much that’s reasonable. If we’re going to shove more air molecules around, we need to also have more diaphragm area.

One of the best tidbits I got from my conversation with Mr. McNutt was in regards to the advantage of using a squared term instead of a linear term. Doubling a driver’s excursion (the linear term) certainly gets you something, but doubling the driver radius (the squared term) gets you much more.

For the sake of argument, let’s simplify a loudspeaker driver’s diaphragm into being a piston that pushes hydraulic fluid around. We’ll conveniently use a driver that starts with 1 mm of excursion, because it will make the math easier. My guess is that most compression drivers can handle rather less excursion than that, but this is just an example. The radius will be 25.4 mm (that’s like a 2″ diameter compression driver, if you want to visualize it).

Displacement Volume = Area * Excursion

Displacement Volume = (pi*25.4mm^2) * 1mm = 2027 mm^3

If we double the linear term to 2 mm of excursion, the displacement doubles to 4054 mm^3. Nice, but if we double the squared term and leave the excursion alone:

Displacement Volume = (pi*50.8mm^2) * 1mm = 8107 mm^3

That’s a fourfold increase in the amount of fluid the piston moved. When it comes to loudspeakers, making a small driver have a very long excursion is impractical, but making a driver with a larger surface area is commonplace. So, if we consider an 18″ diameter subwoofer (228.6 mm radius) that can handle an excursion of 8 mm:

Displacement Volume = (pi*228.6mm^2) * 8mm = 1313386 mm^3

That’s 648 times more displacement, gotten mostly by making the driver bigger.

I can’t say exactly how all this works out with real drivers, real air, and the real equation for intensity. However, even with rough approximations it seems pretty clear that it’s much easier to move a lot more air if you have a big diaphragm available. The squared term is very important in getting the necessary results.

The Best Upgrades

If you’re going to upgrade something, try to upgrade at the ends of your signal chain.

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This business is so “magical gear” oriented that it hurts people. I don’t know how many bankruptcies, strained relationships, failed businesses, and heartburn prescriptions have resulted from gear acquisition, but my bet is that the number is somewhere between “a lot” and “a gazillion.” Audio humans spend a ton of money, and what’s worse, there’s a tendency to spend it on the wrong things. The search for better sound is a journey that’s often undertaken through a path that leads into the deep underbrush of mythology, and that’s a recipe for getting lost.

One perennial (and expensive) mistake is pursuing upgrades to the wrong parts of the signal path. Folks get incredibly wound up about the sound quality of things like consoles, poweramps, preamps, and even cables. They thrash around, trying to figure out why things don’t sound “just so,” and run huge bills as they do. In the process, they miss opportunities to upgrade the bits that would really matter.

If we’re talking about the part of the signal chain that involves electricity, the bits that matter are at the ends.

Transduction Is Hard

Let’s start with what I’m not saying: I’m not saying that the middle of the signal chain is trivial. It isn’t. A lot of work has been done to get us to where we are now in terms of distortion and SNR. Very smart people have worked for decades to design and miniaturize the components and subassemblies that make pro-audio go. What I am saying, though, is that signal routing, combining, and gain adjustment ARE trivial when compared to signal transduction.

For instance, let’s take the INA217, an instrumentation amplifier that can be used to build microphone preamps. At around 68 dB of gain, (the base 10 logarithm of 2500, multiplied by 20), the unit maintains a bandwidth beyond the audible range. Nifty, eh?

You can buy one for less than $7. Buy in quantity, and the per-unit cost is less than half that.

Or, take a mix bus from a console. The heart of a mix bus is either electrical or mathematical summing. Addition, I mean. The basic process is incredibly simple, and though the circuits do have some important particulars, they are not difficult for an electrical engineer to design. (And, that’s assuming that they actually get designed anymore. I strongly suspect that most folks are grabbing an existing design from a library and extending it to meet a certain specification.) Insofar as I can determine, there is no secret sauce to a summing bus. There are better components that you can specify, and due diligence is required to prevent external noise from corrupting the signals you actually want to use, but there’s no “magical addition process” that some folks have and some don’t.

“Doing stuff” to electricity that’s already electricity is pretty darn simple.

Life gets far more complicated when you’re trying to change sound into electricity or back again. The vagaries of directional microphone tuning, for instance, are strange enough that they don’t even make it into patent applications. They’re kept locked away as trade secrets. Microphone diaphragms aren’t really something you can build with ingredients found in your kitchen (good luck with working on materials that are only microns thick). Just about any decision you make will probably affect the whole-device transfer function in a way that’s easy to hear. On the output side, the tradeoffs associated with making a loudspeaker driver are both numerous and enormous. Everything matters, from the diaphragm material on up. The problem compounds when you start putting those drivers in boxes and attaching them to horns. Big drivers move lots of air, but don’t start or stop as fast as small units. The box might be resonating in a strange way. Just how bad do things get when the loudspeaker is run below the box tuning? Again, a small design change is likely to have audible results.

Manufacturers continue to iterate on transducer designs in ways that appear “fundamental” to the layman, whereas iteration on other products is more about incremental improvements and feature additions.

What this all amounts to is that a transduction improvement is far more likely to be of obvious and significant benefit than an upgrade in the “pure electricity” path.

Beyond The Chain

Upgrading the ends of the signal chain is a concept that works even beyond the electro-acoustical sense.

Let’s say I have the greatest microphone ever made. The entire thing is built from pure “unobtainium.” It is perfectly linear from 1 Hz to 30 kHz, and has infinitely fast transient response. It’s not even physically possible for this microphone to exist, it’s so good. I put that microphone in front of a singer with an annoying overtone in their voice. Does that singer sound good?

No. The microphone perfectly captures that ugly harmonic. If I had a choice, I would prefer an upgrade to the ultimate end of the signal chain: The signal source. I’ll take an amazing singer into an okay mic at any time, but a great mic in front of a bad singer doesn’t help very much.

Let’s also say that I have the greatest loudspeaker ever constructed. Its transfer function is perfectly flat, with flawless phase response. This mythical device is then placed in an aircraft hangar built of metal. The acoustical environment’s insane reflections and smeared transients result in a sound that’s almost completely unintelligible, and even a bit painful.

A “basically okay” loudspeaker in a great room would be much better.

If you’re going to undertake some sort of sonic improvement, you want to do all you can to upgrade things that are as close to the endpoints as possible. If you’re not getting the sound you want, look at source quality, room acoustics, mic capability, and loudspeaker fidelity first.