Tag Archives: System Building

Pretty Close To An SC48

The great thing about this business is that, nowadays, you can get a lot of functionality for a little money.

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 just had the privilege of spending four days working in an amazing venue. I won’t be naming names here on the site, although where I worked (and who I worked for) are not secret – it’s just a bit of courtesy, as it was my “first rodeo” with the performing group.

The venue was not small by my standards. A 500-seater sits squarely in what I consider the “midsize” bracket. Also, the place has a gloriously high ceiling, full fly-system (you know, curtains, big battens to hang lights on, that sort of thing), tons of power, and pretty much whatever else you want. Plus, they have a helpful, good-natured, knowledgeable staff that are always around when you need them.

At FOH, they installed an SC48.

An SC48 is a tour-grade digital console by Avid. It’s one of those pieces of gear that folks salivate over, and with good reason. It’s got an eminently usable control surface, a well-designed software interface, and lots of channels. Plus, as I said, it is an honest-to-goodness tour-grade unit. When you’re driving one, you are very definitely sitting in “the big chair.”

A basic model of the SC48, purchased new, will run you about $29,000 US.

And, for less than 1/10th of that, you can buy a digital console that will basically do all the same things an SC48 can do.

I’m Not Slagging The SC48

I hope that it’s abundantly clear that I am in no way ragging on the Avid product. There are things that I wish were different on it, but that can hold on for a bit.

What I am saying is that the gap between “pro-sumer” units and the biggest, coolest toys is continually narrowing.

See, I have in my possession, right now, a Behringer X32. It’s not even the full-size model. Spending four days with an SC48 made it very clear to me that an X32’s core functions are entirely competitive with the Avid desk. By extension, this means that pretty much any “affordable” digi-mixer is competitive on the basis of core functionality.

Full dynamics processing available on all input channels? Check.

Multi-band, fully parametric EQ on all input channels? Check.

A snapshot system? Check.

Recallable input gains? Check.

Matrix mix functionality? Check. (Matrix mixing is creating a blend of inputs and/ or outputs, as opposed to regular bus and aux mixes which are input-fed only. I don’t really use matrices, but it is one of the features, so…)

Now, let’s be fair. When you invest in something like an SC48, you’re buying more than just the core functionality. You’re buying (hopefully) great manufacturer support, which can get you out of a jam on nights and weekends. You’re buying redundant power supplies. You’re buying industry recognition and acceptance of the hardware and software platform. You’re buying (again, hopefully) better and more careful manufacturing. You’re buying a product which is meant to have a lengthened life cycle.

None of that is a mere triviality.

At the same time, though, those elements represent a VERY large price premium that doesn’t really make sense for small-venue types.

How Much Is It Worth To You?

Yes, an SC48 can run ProTools plugins, which is something my X32 can’t handle.

I did find that functionality very useful!

Because – for some bizarre reason – Avid doesn’t seem to think that integrated dynamics and EQ on OUTPUT channels is something anybody needs. (Avid…guys…if a console costs as much as a car, I really think that full processing on outputs ought to be there. Just an idea. Behringer can help you with that, as can Soundcraft, A&H, Yamaha, whoever you like.) Also, an X32 can’t crossfade from scene-to-scene, whereas an SC48 does it intuitively and effortlessly. Along with that, there’s very finely-grained control over what is “recall safe” on the Avid. I liked all that for the show I was doing, and it’s super-nifty in general, but I don’t know if I’d be willing to pay $26,000 extra for the privileges.

The ease of patching on the Avid unit blows most other implementations completely out of the water. Again, though, I’m not sure that’s worth a 14X price differential. (As a side note, if you can handle the routing matrix in Reaper, you can patch on an SC48. The concepts are exactly the same.)

Pretty much the only thing that you can’t get around is the option of having 48 inputs in one frame.

I realize that this sounds dangerously close to ripping on the SC. What it really is, though, is a celebration of just how level the playing field is becoming. Some folks lament that everything is turning into software; I, on the other hand, think it’s great. It means that affordable gear has staggering power and flexibility. The work you can do with a relatively inexpensive mixer really is not that far away from what a big-time desk can pull-off. There are definitely folks who need the tour-grade units, and can pay for them. You HAVE to have the appropriate tool for the job, and I’m not suggesting that folks who need all that an SC48-class console provides should use an incorrect tool.

I’m just saying that, more and more, the technological barriers to the best possible sound being available from a console are collapsing. As time goes on, operator dedication, curiosity, and professionalism – which have always mattered the most, anyway – are completely eclipsing the limitations of the “toolkit.”

Because the toolkit is getting better and more capable on a continuous basis.


All Powered Speakers Are Not Created Equal

March’s Schwilly guest post.

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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“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.


Maybe The Only Way Out Is “Thru”

Out may be “thru,” but “thru” usually isn’t out.

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 labeling of jacks and connections is an inexact science.

Really.

For instance, there are audio devices with “in” and “out” jacks where you can connect a source to either point and be just fine. It might be confusing, though, to have two areas labeled “input” (or even “parallel input’), so one jack gets picked to be “in,” with the other as its opposite.

At some point, you just get used to this kind of thing. You trundle along happily, connecting things together without a care in the world.

…and then, somebody asks you a question, and you have to think about what you’re doing. Just why is that jack labeled as it is? You’re taking signal from that connector and sending it somewhere else, so that’s “out,” right? Why is it labeled “through” or “thru,” then?

The best way I can put it to you is this: Usually, when a manufacturer takes the trouble to label something as “thru,” what appears on that connector is the input signal, having gone through the minimum necessary electronics to make the connection practical and easy to use. A label that reads “out” may be a signal that passed through a lot of electronics, or it may be a “thru” that’s simply been called something that’s easier to understand.

“Thru,” From Simple To Complicated

thru-wire

That up there is a simplified depiction of the simplest possible “thru.” It’s two connection points, with nothing but some sort of conductive connection between them. Also on that connection is some sort of internal arrangement of electronics. In this kind of thru, you might see male and female jacks on the different points (if the connections are XLR), but the reality is that both connectors can work for incoming or outgoing signals. Put electricity on either jack, and the simple conductors between those jacks ensure that the signal is present on the other connection point.

This kind of thru is very common on passive loudspeakers and a good many DI boxes. You might see a connector that says “in,” and one that says “out,” but they’re really a parallel setup that feeds both an internal pathway and the “jumper” to the other connector. Because the electrical arrangement is truly parallel, the upstream device driving the signal lines sees the impedance of each connected unit simultaneously. This leads to a total impedance DROP as more units are connected; More electrical pathways are available, which means lower opposition to current overall.

thru-buffer

So, what’s this, then?

This is a buffered thru. In this case, the two jacks are NOT interchangeable. One connector is meant to receive a signal that gets passed on to internal electronics. That connector is linked to a jack with outgoing signal, but in between them is a gain stage (such as an op-amp). The gain stage probably is not meant to perform meaningful voltage amplification on the input. If two volts RMS show up at the input, two volts RMS should be present at the output. The idea is to use that gain stage as an impedance buffer. The op-amp presents a very high input impedance to the upstream signal source, which makes the line easy to drive. That is, the buffer amp makes the input impedance of the next device “invisible” to the upstream signal provider. A very long chain of devices is made possible by this setup, because significant signal loss due to dropping impedance is prevented.

(Then again, the noise floor does go up as each gain stage feeds another. There’s no free lunch.)

In this case, you no longer have a parallel connection between devices. You instead have a serial connection from buffer amp to buffer amp.

thru-logic

The most sophisticated kind of thru (that I know of) is a connection that has intervening logic. There can be several gradations of complexity on that front, and a “thru” with logic isn’t something that you tend to see in audio-signal applications. It’s more for connection networks that involve data, like MIDI, DMX, and computing. The logic may be very simple, like the basic inversion of the output of an opto-isolator. It can also be more complex, like receiving an input signal and then making a whole new copy of that signal to transmit down the chain.

A connection this complex might not really seem like a “thru,” but the point remains that what’s available at the send connection is meant to be, as much as possible, the original signal that was present at the receive connection…or a new signal that behaves identically to the original.

Moving Out

So, if all of the above is “thru,” what is “out?”

In my experience the point of an “out” is to deliver a signal which is intended to have been noticeably transformed in some way by internal processing. For instance, with a mixing console, an input signal has probably gone through (at the very least) an EQ section and a summing amplifier. It’s entirely possible to route the signal in such a way that an input is basically transferred straight through, but that’s not really what the signal path is for.

With connection jacks, the label doesn’t always tell you exactly what’s going on. There might be a whole lot happening, or there might be almost nothing at all between the input and output side. You have to look at your owner’s manual – or pop open an access cover – to find out.


For The Love Of Trim Height

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

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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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

lowtrim

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.

highertrim

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?

hightrim

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.

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 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.

pileflat

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.

pileoneoff

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.)

piletwooff

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:

oneboxlate

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.

twoboxeslate

The other one is 3 ms late.

threeboxeslate

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.


I Am SO Over Wireless

Another “Schwilly” article.

Please Remember:

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

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“If you are an audio person or a musician, someone you know will eventually want to do things involving audio (or data representing audio) and radio waves. They will think that such an idea is brilliant. They will think it will be so very nifty to be un-tethered and free, wild like the stallions and mares which once loped across the mighty plains of America’s central expanse, majestic in their equine kingship ov-

Yeah. About that. Don’t believe it. Wireless is a pain in the donkey.”

Read the whole thing for free at Schwilly Family Musicians.


Deterministic Troubleshooting

Audio systems can be counted on to be deterministic in every practical way, but you can’t always count on them to be the same system.

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.

In my experience, audio systems are deterministic creatures at the practical level.

That is to say, giving the system an input results in getting an output which has always been transformed in the same way. Introduce an input into a perfectly linear system which runs through a +6 dB gainstage, and you can count on getting your input, +6 dB, at the output. Give your input to a system with a nonlinear transfer function and a +6 dB gainstage, and you can be confident that the other end of the system will spit out your input signal, plus the filtering of the transfer function, and +6 dB of level.

I am, of course, excluding noise.

Electronic self-noise, if you’re being really picky, displays “random” behavior. The overall level and frequency content is statistically predictable, but the exact output at any given instant can’t be precisely foreseen. If you include the noise in your description of the output, then yes, you can argue that a sound system is non-deterministic.

But, let’s be real. At a mental level, whether we’re listening to a system or describing its behavior, we subtract the noise. There’s also the whole matter of best practices dictating that systems should be run in such a way that the noise is insignificant when compared to the signals, which makes that mental subtraction trivial.

Anyway – as far as I can tell, audio systems are deterministic in every way that really matters.

This, and a somewhat surprising component of it, can inform our concepts of troubleshooting.

Apparent Non-Determinism Makes Life Difficult

Notice that I said, “apparent.”

See, I was originally going to write this article by saying that a sound rig displaying non-deterministic behavior is one of the toughest things to troubleshoot. However, in thinking about things more fully, I’ve come to the conclusion that audio setups aren’t really capable of being non-deterministic. They can SEEM to become non-deterministic, but they aren’t actually. What truly happens is that you expected one particular system, but actually have something else.

Let’s say that you set up a FOH PA system. Everything is working nicely. You give it an input, and you get the output you expect from the loudspeakers. Groovy.

Then, when you’re not looking, somebody accidentally pulls the connection that links the console with all the downstream bits. Unaware of this, you return to FOH control and give the system an input. The output is silence. It’s not the output you expected, but the system hasn’t entered a non-deterministic state. Rather, what you have is a different system: A system that ends at the console. A mixing console that can’t pass its signals to an output transducer has an acoustical gain of 0. (Not 0 dB! A gain of 0. The signal is multiplied by 0. Any signal multiplied by 0 is 0.) The system, in truth, is both entirely deterministic and highly predictable. Any input to it results in no acoustical output.

So, no, the rig isn’t non-deterministic – it’s just not the system you expected to be in place. There is a reason that the setup is not behaving as expected, and that reason (a disconnected cable, in this case) can be isolated and corrected.

I recognize that this is small comfort to those of you, including me, who have been faced with intermittent problems. Intermittent failures are the hardest to fix, and the most nail-biting to endure, because you’re up against something that APPEARS to be non-deterministic. The output from a given input becomes unpredictable. You might be just fine for hours, and then things go very wrong, and then you’re fine for a few minutes, and then something else happens, and your blood-pressure goes through the roof.

Basically, you’re trying to debug multiple systems without knowing which system you’re currently debugging. (Good luck with that.) This is why I have been known to breathe a sigh of relief when a rig “breaks,” and has the decency to then STAY broken. When that happens, I only have to troubleshoot one system; A deterministic system which either has no output, or some output which is obviously and predictably not what I want.

If you’re faced with an audio setup that appears to be doing things at random, what you really have is – very likely – more than one system. Those systems are trading places at unpredictable times. Every one of those systems is deterministic, so what you have to do is stop all systems but one from manifesting themselves. If you can directly get that one system to be the setup that gives you the desired output, that’s great. If you can’t, though, that’s really okay. Even if the output isn’t what you ultimately want, causing your assemblage of gear to settle into one stable state will give you the leverage necessary to figure out what’s wrong and fix it. If the system seems to be non-deterministic, do what you have to do to make it appear to be as deterministic as it actually is. You’ll be back in control, and control is what you need in order to get things working again.


VRX Brackets

One word: No.

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.

vrxWant to use this image for something else? Great! This picture is a derivative of this file by JPRoche on Wikimedia Commons.

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.)


Impedance River

Good luck with trying to fill the Mississippi using a fish-tank pump.

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.

Yup – there are a million-billion analogies about impedance. I’m going to do one more anyway. Maybe this will be the one somebody reads – you never know.

The electrical underpinnings of this business range from simple concepts to alien geometries. Looking at the “EE” side of audio is a great way to realize just how much you don’t know. It’s particularly tough on non-technical folk and newbies, for whom the “alien geometries” bar is very low. They’re trying to get things all plugged in and playing nicely, but they’re also faced with these mysterious pronouncements on rear panels: “Minimum Load: 4 Ohms.”

Seriously…what do you mean, “minimum?” A load is something you can carry, and you want to avoid having too much, right? WHAT INSANITY IS THIS?

Anyway.

Impedance is an important topic for those of us who use electricity to create various noises. I personally think that the basics ARE fairly intuitive, but are rarely presented in an intuitive way. So, I’m going to do my best.

Up A Creek

Water and electricity are very analogous to one another. Voltage is like pressure, and current is like, well, current. Flow, you might say. The amount that’s “going by” per unit of time. Impedance is opposition to flow in an AC (Alternating Current) circuit. This opposition to current varies with frequency. Some circuits have very high impedance at low frequencies, but pass high frequency material readily. Other circuits do the reverse. Other circuits easily pass a specific range of frequencies, and offer higher impedance on both the high and low sides of that range. (This is how you make analog EQ, by the way.)

Rivers and streams are imperfect analogies, because they are really examples of DC (Direct Current). Opposition to DC flow is resistance, and it’s much simpler than impedance overall. Nevertheless, simple is a good way to start.

Consider two waterways. One is a creek. The creek is seven feet wide and five feet deep. The other is a major river that’s 500 feet across and 50 feet from the surface to the bottom. Here’s an SVG to help you visualize the difference in scale:

creek_vs_river

What if you could get 25,000 cubic feet of water to flow down both waterways each second. Which one would be likely to knock you off your feet and slam you into a rock?

The creek, of course.

The creek has a very small cross-section when compared to the river. In order to get 25,000 cubic feet of water down the creek every second, the fluid would have to flow at a speed of 714 feet per second. That’s about 487 MPH. (!) The big river, on the other hand, is going less than 3/4 of a mile per hour. The narrow, little creek offers a proportionally high opposition to flow, so getting the same amount of current as the river requires a lot of pressure – or voltage, if electricity is our thing.

What you can begin to see here is the relationship described by Ohm’s law. If the impedance of a circuit rises, maintaining the same flow requires greater “motive force/” pressure/ voltage. If the impedance drops, maintaining the same voltage creates more flow. (If you could run a main sewer pipe at the same pressure as a power-washer, you would have a LOT of water going down that pipe.)

What This Means To You

Let’s say you have a pump. It’s built for a home aquarium. It has no trouble at all pressurizing a 1/8″ tube and keeping water flowing along.

Now…connect that pump to a large sewer pipe, and try to pressurize THAT.

Good luck.

Even if you did something rather dangerous (do not try this at home, or AT ALL) and found a way to make the pump run harder, all you’d do is burn out the poor thing. The unit simply wasn’t designed to put that kind of flow down a pipe.

This basic principle is why amplifiers have “minimum” load ratings. Loudspeakers connected in parallel are effectively being attached as a larger and larger “pipe.” The overall circuit impedance goes DOWN, because there are more possible paths for the electricity to take. It becomes easier and easier for electricity to flow somewhere. The problem is that your amplifier is a pump that attempts to create a constant pressure. That is, if you have one load attached, and you send an input signal that should result in, say, 50 VRMS (Volts RMS) at the amplifier outputs, the amp will attempt to swing 50 VRMS at the outputs if you change the load. To fill the larger pipe, the amplifier has to supply proportionately more energy to maintain the voltage.

At some point of decreasing load impedance, the amp just can’t keep up. It can’t deliver that much energy on a continuous basis. It’s running hotter and hotter, with greater stress on everything from the power supply to the output devices. Eventually, depending on the amp’s sophistication, it might “thermal” and shut itself off, current-limit by throwing a resettable breaker, or drop its output to keep giving you something whilst recovering.

This also connects to the whole issue of impedance bridging, which is what enables maximum voltage transfer. Maximum voltage transfer is what we want in pro-audio, and it happens when low output impedances drive high INPUT impedances. To keep the water analogy going, it’s like connecting a city water line to a house. The city water is a big pipe (low impedance), which feeds the rather smaller house inlet (high impedance). As long as everything is working correctly, the city line has no trouble keeping the house inlet fully supplied. There’s plenty of pressure available to the house, because of good pressure (voltage) transfer.

The opposite of this is when you connect something like a piezo pickup to a “vanilla,” passive DI box. The piezo transducer actually makes a good bit of pressure, but its output impedance is very high. It’s like one of those tiny little capillary tubes that the folks at the doctor’s office use when drawing blood from a finger stick. The input impedance of the passive DI is actually pretty high, but it’s proportionally low compared to the piezo output impedance. The piezo drives the passive DI poorly, resulting in very low level, and the circuit configuration causes a noticeable loss of low-frequency material.

Solve the impedance bridging problem by connecting an active DI with very high input impedance, and your problems go away.

The overall point is that you can’t fill an infinitely large conduit with a finite supply. That’s why audio devices have appropriate loads for their outputs, and why you have to be mindful of those loads.


Buzzkill

Ridding yourself of hum and buzz is like all other troubleshooting: You have to isolate the problem to fix it.

Please Remember:

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

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

Not all hums and buzzes are equally bad. Honeybees hum and buzz, but they’re super-helpful creatures that are generally interested in being left alone and making honey. Wasps, like the one pictured above, are aggressive jerks.

Of course, this site isn’t about insects. It’s about audio, where hum and buzz mean problems. Unwanted noise. Blech.

I recently got an email from a friend who wanted to know how to de-buzzify (I just made that word up) a powered mixer. When you mercilessly distill what I told him, you come up with a basic truth that covers all of troubleshooting:

The probability of an effective fix for a problem is directly proportional to your ability to isolate the problem.

Solitude

The importance of finding the exact location of a fault is something that I don’t believe I can overemphasize. It’s the key to all the problem-solving I’ve ever had to do. It doesn’t matter if the problem is related to audio signal flow, car trouble, or computer programming; if you can actually nail down the location of the problem, you’ve got a real shot at an effective (and elegant) fix.

The reverse is also true. The less able you are to pinpoint your conundrum’s place of residence, the more likely you are to end up doing surgery with a sledgehammer. If you can’t zero-in on a root cause, you end up “fixing” a certain amount of things that aren’t actually being troublesome. The good news is that you can usually take an iterative approach. All problems begin with “this system isn’t working as I expected,” which is a completely non-specific view – but they don’t have to end there. The key is to progressively determine whether each interrelated part of the system is contributing to the issue or not. There are lots of ways to do this, but all the possible methods are essentially an expression of one question:

“Is the output of this part of the system what I expect it to be?”

So…here’s a way to apply this to buzz and hum problems.

Desperately Seeking Silence

Talking in depth about the exact electrical whys and wherefores surrounding strange and unwanted noises is a little bit beyond my experience. At a general level, though, the terminology of “ground loop” provides a major clue. Voltage that should be taking a direct path to ground is instead taking a “looping” or “circuitous” path. A common cause of this is equipment receiving mains (“wall”) power from two different circuits, where each path to mains ground has a significantly different impedance. There is now a voltage potential between the two pieces of gear.

Bzzzzzzzz….

You can also have a situation where two device’s audio grounds are interconnected such that there is a potential between the two devices.

Hmmmmmmzzzzzzzz…

Anyway.

The first thing to do is to decide what piece of equipment you’re testing against. Maybe it’s a mixing console. Maybe it’s an amplifier. Whatever it is, you are asking the question from before:

“Is the output of this part of the system what I expect it to be?”

Or, more specifically…

“I expect this device’s output to be quiet, unless an audio signal is present. Is that the case?”

To answer that question, you need isolation.

WARNING: At NO point should you do anything to disconnect the mains-power/ safety grounds from your equipment. It’s there to prevent you from dying if the equipment chassis should become energized. In fact, as a start, try to verify that the mains-power sockets you are using actually DO provide a connection to “earth.” If they don’t, stop using them until they’re fixed. You may even find that your noise problem goes away.

To get isolation, start by disconnecting as much as you possibly can from the DUT (the Device Under Test). Of course, you’ve got to have some kind of way to monitor the output, so that might mean that you can’t disconnect everything. As much as possible, try to ensure that all mains-power grounds offer the same impedance – if it must stay connected, and it requires mains power, get all the power to connect to the same socket. A multi-outlet power tap can come in handy for this.

Is the output what you expect?

If yes, then something which was connected to your DUT’s input has a good chance of being the problem. At this point, if possible, treat each potential culprit as a secondary DUT in turn. If feasible, connect each suspect directly to your monitoring solution. If the ground loop manifests itself, and the suspect device requires mains power, try getting power from the same tap that the primary DUT is on. If the loop goes away, you’ve established that the two devices in play were likely having an “unequal impedance to ground” problem. If the loop stays in effect, you can jump back up to the beginning of this process and try again, but with the gear you had just plugged in as the new, primary DUT. You can keep doing this, “moving up the stack” of things to test until you finally isolate the piece of gear that’s being evil. (IMPORTANT: Any piece of the chain could be your problem source. This includes cables. You may need to pack a lunch if you have a lot of potential loop-causers to go through.)

If you can’t get the buzz to manifest when adding things back one at a time, then you might have a multi-device interaction. If possible, work through every possible combination of input connections until you get your noise to happen.

But what if the output on the original DUT was NOT what you expected, even with everything pulled off the output side?

At that point, you know that an input device isn’t the source of your trouble with this particular DUT. This is good – your problem is becoming isolated to a smaller and smaller pool of possibilities.

Try to find an alternate way to connect to your monitoring solution, like a different cable. If the problem goes away, that locates the cable as the menace. If you’re switching the connection, and the noise remains with no audio path, then the monitoring system has the problem and you need to restart with a new DUT. (If you’ve got a mixer connected to an amp and a speaker, and a ground loop stays audible when the mixer-to-amp connection is broken, then the amp is your noise source.)

If you’ve tried all that and you still have the buzz, it’s time to try a different circuit. Get as far away from the original mains-power socket as you can, and reproduce the minimal setup. If the ground-loop goes away, then you may have a site-wiring issue that’s local to the original socket(s). If the problem doesn’t go away, it’s time to take a field-trip to another building. It’s possible to have a site-wide electrical problem.

If the loop still won’t resolve, it’s very likely that your DUT has an internal fault that needs attention. Whether that means repair or replace is an exercise left to the reader.

Hopefully, you don’t get to that point – but you won’t figure out if you ARE at that point unless you can isolate your problem.