Tag Archives: System Building

How To Buy A Microphone For Live Performance

A guest-post for Schwilly Family Musicians

Please Remember:

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

vintage_microphone-wallpaper-1280x800

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


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


How Much Light For Your Dollar?

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

Please Remember:

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

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

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

Of course, that means video.

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

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

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

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

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

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

The Experiment

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

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

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

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

Step 5: Darken the hallway.

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

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

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

Step 9: Note the value indicated by the meter.

Important Notes

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

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

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

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

The calculation for 1 meter lux was as follows:

19′ = 5.7912 m

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

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

Multiply 19′ lux by 4^2.53 (33.53)

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

The Data

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


Ascending sort by:


Why I’m Excited About The New X32-Edit

Alternative interfaces are best when they actually leverage the power of being alternative.

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.

x32edit-screenWant to use this image for something else? Great! Click it for the link to a high-res or resolution-independent version.

Even if you don’t use X32-Edit, the remote/ offline software for Behringer’s X32 series of consoles, I think you should keep reading. I say this because the point of this article is not to “dig deep” into the feature set of X32-Edit. Rather, I want to speak in (fairly) general terms about what console-remote software can get right, and not so right.

So, anyway…

I’m a publicly avowed fan of Behringer’s X18. I’m especially a fan of the control software, which I feel absolutely nailed what console control software should be. The ironic thing was that I felt the X18 application was markedly BETTER than the remote control/ offline editor for the X32 – and the X32 is the higher-tier product!

But why would that be?

Well, rather like the gentlemen of “Car Talk,” I have a theory – or, more correctly, a hypothesis. My guess is that the X18 software was better because it was free, from the very beginning, to act purely as a virtualized interface. The X32 series is solidly founded on consoles which have a real control surface, the only true exception being the X32 Core model. An X18 and its cousins, on the other hand, are built on the idea of having almost no physical controls at all.

With the X32, then, it was very easy for the software designers to choose to closely emulate the look and feel of the physical control surface. In the case of the X18, there was never any surface to copy – and the control implementation benefited greatly as a result. The software was always meant to be a connection to something abstract; DSP and digital console commands have no physical form that they are required to take. With this being the case, the presentation of the controls could be built to fully embrace the nature of a display device fundamentally decoupled from the console. The control layout can be rearranged to best leverage whatever screen size and geometry is available. Actions can be streamlined, contextualized, and made more powerful with the recognition that a user can apply multiple control gestures (click, long-click, double click, right-click, etc) on a single element. You can easily have a console overview that provides a ton of information, yet remains interactive.

The X18 software took great advantage of the above, which meant that I immediately recognized it as the way that X32-Edit SHOULD have worked. To be both clear and fair, the previous iterations of X32-Edit weren’t poor or unusable. What they were was “conflicted.” They sort of took advantage of what a large, decoupled view device could do for console usage, but they also often limited their behavior based on the limitations of the physical control surface’s display. Why make something less capable than it can be? In my mind, yes, there is a point in having familiarity – but getting powerful usage out of a console is more about understanding the concept of what you want to do than memorizing the button presses to do it.

Also, the old X32 remote implementation never showed as much overview as it could have with all the screen real-estate that was available, and it couldn’t really “flow” itself into different screen shapes and resolutions either. It had a basically fixed size and aspect-ratio, and if that didn’t take advantage of what was there…tough.

Thus, I am very, very happy with the new X32-Edit. It acts like a beefed-up version of the X18 application, taking all kinds of advantage of being a virtual window into the mixer. Everything seems to be more immediately accessible, and the display offers real customization in terms of what you’re looking at. The software isn’t trying to be a copy of the control surface; It’s trying to be a replacement for it.

And that has made X32-Edit into the software that it always should have been.


The Pros And Cons Of Distributed Monitor Mixing

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

Please Remember:

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

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

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

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


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

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

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

Con: Monitors Become Harder For The Player

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

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

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

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

Con: Not Using In-Ears May Cause Trouble

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

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

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

Pro: Integrated Ecosystems Are Powerful And Easy

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

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

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

Pro: Everybody Can Have Their Own Mixer

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

Con: Everybody Having Their Own Mixer Is Expensive

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

Pro: Personal Mixing Is Luxurious

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

Con: Not Everybody May Buy In To The Idea

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

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

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


Double Hung Discussion

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

Please Remember:

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

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

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

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

Why Do It At All?

First, we can start with that logistics thing again.

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

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

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

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

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

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

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

System Output And Response

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

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

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

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

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

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

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

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

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

combfiltering0.22ms

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

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

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

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

More On Output – Enough Rig For The Gig?

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

The people with that concern are entirely correct.

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

The Raw Numbers

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

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

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

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

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

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

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

The Specifics Of The Show

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

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

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


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


The Story Of A Road Gig, Part 3

Commentary with pictures – or maybe it’s the other way around.

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.

road-gig-3Want to use this image for something else? Great! Click it for the link to a high-res or resolution-independent version.
Rather than try to relate the entire story of this overnighter as a narrative, I’ve decided to take the approach of commenting on the various photos that were taken at the gig (or around the process of it). There are, amazingly, some shots where yours truly makes an appearance. Scotty of Eyes Open got ahold of my camera, and, well, there ya go.


X32 Cores

X32 Cores

While it’s not necessarily for the faint of heart, running surfaceless consoles can potentially save you money, weight, and some space. Consoles like this really hammer home what a digital mixer is: A whole lot of software running on specialized hardware. Delete the control hardware, and all the heavy-lifting for audio still remains.

Going surfaceless requires significant homework. You’ll have to get both your “mix brains” and their associated control devices (laptops, tablets, etc.) onto a network and talking to each other. An inexpensive wireless router is really all you need for this, but DO have a fallback option. Also, anything that doesn’t need to be wireless probably shouldn’t be, so use a wired connection to your control gear whenever you can. Ethernet cable is cheap, available almost everywhere, and pretty much stupid-proof.

And, for heaven’s sake, set up meaningful security on your wireless network. Nothing but your consoles and controllers should be connected to it.

I have two X32 cores for more than one reason. Reason #1 is to be able to have separate FOH and monitor worlds with full “first-class” channel counts – 32 inputs each. Reason #2 is that, if one console were to give up the ghost, I could fall back to its counterpart and keep going.

As much as is practical, build mix templates for your show before you leave. The ability to walk up to the show and “just go for it” without having to think through everything on the fly is a big help. Remember to do some meaningful tests on your setup to ensure that it works, and that you know how it works.

S16 Stageboxes

S16 Stageboxes

Digital stageboxes help you save space and weight by removing the need for a big, heavy, multicore trunk. The irony is that digital stageboxes are rather more expensive than their analog cousins. Your overall cost may be slightly reduced if you get a single unit with all the inputs and outputs you need, but you have to account for the risk of that unit dying on you. Using two boxes to do the job allows you to continue in some way if one of them stops cooperating.

Use the network cabling recommended by the manufacturer. If your digital snake system calls for shielded Cat5e with Ethercon connectors, that’s what you should use. There are plenty of stories out there of people who encountered…interesting results while using connectivity that was not up to spec. (At the same time, I’m not convinced that “super premium” is necessary. GLS Audio makes SSTP ethercon cables that seem pretty darn good, and clock-in at under $1.00/ foot.)

Remember to have spare cables for this high-speed, highly-mission-critical audio network you’re building.

Which One Is Which?

Which One Is Which?

Here we see a common, North-American noise-louderizer with a remote console control, he being somewhat perplexed by how the mix-bus order is now reversed due to his move from FOH to the stage.

Tablets And Monitors

Tablets And Monitors

I am brand new to the whole idea of walking up on deck with a remote, but let me tell you, it’s one of the greatest things since sliced bread. For your initial rough-in of monitor world, it’s downright beautiful to be able to put things together without any guesswork, or running back and forth to a console. Instead, you park yourself in front of a wedge, start dialing things up, and instantly hear the results of your changes. This means that you can actually pick up on the exact point where additional gain on a channel starts to get “weird.”

It’s also beautiful to have the remote when artists are actually on stage. Again, a lot of guesswork and disconnection simply goes away. You can talk to each other naturally, for a start. Even more important, though, is that you can actually hear what the musician is hearing. Problems with a mix don’t have to be described, as you can experience them directly for yourself. Finally, it’s a great bit of “politics;” Musicians who have often dealt with uncaring (or just absent) audio-humans now have one who’s really paying attention – and who’s also very much in the same boat as they are.

As was jokingly mentioned above, you do have to remember that your mix order may be “flipped.” If you numbered your mixes based on how you’re looking at things from FOH, walking up on deck now means that you’re seeing the mirror image.

When putting a system together, don’t be stingy with your monitor mixes. I’ve never regretted having more mixes and wedges available. As I’ve said before, and will probably say again, getting everyone happy on deck means a much better experience at FOH. A recipe for success really is making sure that a big piece of your budget goes to monitor world. Give those drummers “Texas headphones” (a drumfill) if at all possible. They tend to like it.

Scotty And McCrae

Scotty And McCrae

Scotty and McCrae were the guys who brought me out on the trip, and on a practical level, the show would NOT have happened without them. McCrae handled a lot of behind-the-scenes logistical elements in real time, making sure that things like shelter, power, and scheduling were actually working.

Scotty joined with McCrae to form my weekend stage crew. It was a little slice of heaven to work with those guys, because all I had to do was describe what I wanted to happen, and then wait a few minutes. The importance of such a crew, that has a can-do attitude and a real sense of humor, can NOT be overstated. I was able to deliver because (and only because) everybody else did their job.

(Also, a huge “Thank You” goes out to Bayley H. for running the event as a whole, for giving Scotty and me a place to sleep, and for chasing down one of those super-rad Honda generators for us. She was juggling about 80 things all weekend, one of those things being the music, and we were very well taken care of.)

Run!

Run!

Spooked by the sudden noise of a band getting comfortable on deck, a black-footed knob-turner (voluminus maximus) bolts for the safety of FOH.

FOH

FOH
FOH 2

I put FOH control on top of the console case, with monitor world off to the side. The laptops are different colors so that I can tell them apart easily when unpacking them. The trackballs are there because, let’s face it, trackpads are fiddly, imprecise, and (to be both blunt and slightly crass) just tend to suck in general.

Another tip: If your primary monitor-world controller has a case, put the monitor control tablet in that same case. It will make things ever so slightly faster and easier at setup.

Talkback is one of the main reasons to have at least one microphone equipped with a switch. Choose where you want talkback to be routed to, latch the console’s talkback control, and then simply flick the switch on the mic when you want to talk.

Laptops (with good batteries) and a UPS are helpful at FOH, because a power failure means that your audio processing and routing stay up. No, there might not be any audio for them to work on, but they’ll be available immediately when you get the power back.

Troopers

Troopers 1
Troopers 2
Troopers 3
Troopers 4

Katie Ainge and her band were real troopers throughout the show. Over the course of two days, we would have a few technical issues, and we would also get rained on twice. Through it all, they played their best, kept smiling, and kept coming back for more:

Originally, they were only supposed to play on the Friday night. However, a storm ended up rolling in. Katie and company played right up until the rain started falling, only calling a halt because their instruments were getting wet. After a hasty pack up and retreat, after which they could have bailed out with full pay, they elected to stay around and get a full show in on the following morning.

Also, large garbage bags make pretty decent rain protectors for loudspeakers and other gear. They do tend to buzz at certain frequencies, but that’s the least of your worries when water starts falling out of the sky.

We only hung a single overhead. With a well-balanced band, a single mic in the right spot will get everything on the kit without getting swamped by bleed. Also, I mix live audio in mono about 99.9% of the time, and a single mic is always in phase with itself.

Try, Try Again

Try, Try Again

After a frantic night of Scotty and McCrae packing, unpacking, and drying out the gear, the next morning came along with the promise of actually doing the show. Notice that the generator really is NOT in the right place. I should have placed it off to the side of the deck, so that the exhaust would have stayed away from the performers. Oops.

Double Hung

Double Hung

McCrae and Bayley, masters of all they survey.

With the PA deployed as it was, putting the same signal into all four FOH mid-highs probably would not have sounded all that hot. The outer pair was slightly behind the inner pair, which would have resulted in the high end being out of phase alignment. That problem did not come into play, however, because the different pairs were used for different signals. The inner pair was my vocal cluster, and the outer pair was for instruments. This technique borrows both from The Grateful Dead’s “Wall Of Sound,” and Dave Rat’s “double hung” PA deployments – it’s just on a very small scale.

The configuration as pictured and described trades coverage area for power and/ or clarity. We essentially have one, larger PA setup that’s firing in a narrow pattern. (Even so, some walking around proved that you could hear the PA pretty much everywhere in the park proper.) An alternative would be to put the entire mix into all four boxes, but aim the boxes to hit different zones. In that case, we’d be trading power/ clarity for coverage.

For Real This Time

For Real This Time 1
For Real This Time 2
For Real This Time 3
For Real This Time 4

With no rain during the actual show, the retry of the previous night went much more smoothly. We did have a couple of problems with the cables to Katie’s DI, with my suspicion being that the metal on their XLR connectors is inexpensive, soft, and therefore prone to change shape when heated significantly in the sun. (I can’t prove it though – this is just a wild theory.)

In any case, though, it was great to see Katie and her friends bring some really enjoyable tunes to an audience able to stay for the duration.

Afterwards, packing the van, we got another rain shower.

But it was time to go home anyway.


Don’t Worry About How It Sounds, Worry About What It Does

Any mixer you buy will sound fine. Pick based on the features and how they work.

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 a console sounds bad – I mean, legitimately and unmistakably – it’s either broken or you’re using it poorly.

(If this post doesn’t kick the hornet’s nest, I will be very surprised.)

My point in this really isn’t to offend. It really isn’t to pick a fight. It really is to be very direct about what to spend your time and worry on when picking out a device to route and combine inputs.

I am by no means the most “well traveled” console operator on Earth. There are guys and girls who have had their paws on many, many more desks than I have, in several thousand more rooms than I’ve been in. At the same time, I have been around long enough to have gotten a pretty good sampling of what’s out there.

I’ve run signals through five-input mini-mixers.

I’ve done “coffeehouse” gigs on ancient monstrosities that I could barely lift. Hugely overgrown beasties which consisted of something like 12 channels, a heavy-as-a-bowel-movement class-AB poweramp (that probably managed a peak output of 400 watts/ side into 8 ohms), knobs and faders that someone with giant hands would have found comfortable, and which had “Peavey” silk-screened on the top surface.

I’ve pushed live audio through consoles that people would be embarrassed to own, and consoles that people would happily show off to some folks, and also through contraptions that nobody could possess but me – because I assembled the thing.

I’ve been on what Avid/ Digidesign would consider a flagship live-mix platform.

I’ve had the opportunity to do real, serious, hands-on, studio-environment stuff with large-frame analog units that would run you about $1,000,000 (in late 1990s dollars) when new.

Let me tell ya, folks,

They all sound basically the same.

Really.

Much like preamps, I have never been in a situation where I thought, “If I just had this one particular console, this would all sound better.” Never.

The Subjective Factor

Some of this has to do with how I work. There are sound craftspersons out there who are into the idea of “special mojo.” The magic of a certain preamp circuit. The plug-and-sweeten behavior of a very specific EQ design. The way the summing bus in a certain piece of signal-combining gear does this beautiful “something” when you hit it just right.

This is all neat stuff. When you’re sitting there, and you’re sure it’s happening, and it’s making your day, that’s great.

It doesn’t generally fit my reality, though. In my world, the time required to find the spot where the snare drum smooshes seductively into the harmonic distortion characteristics of a mic pre is time that would be better spent getting the vocals loud in monitor land. By my methodology, finding a console that gives you some extra forgiveness – or even sounds super-special – when you’re just tickling the overload lights is not a problem to solve. The problem to solve is why your gain structure is messed-up enough to have you bumping into the electrical limits of the desk.

On the flipside, you might be really into this kind of thing, which is fine if it’s working for you and the people around you.

The reason, though, that I point out that I don’t personally find it helpful is for the new folks. The guys and girls who are trying to buy things, and agonizing over spec sheets, scared to death that they’re not going to get enough bang for their buck. The bang is not in those tiny numbers.

What You’re Looking For

What my experience has overwhelmingly shown me over the past years is this: Any console which is basically capable of filling the needs of a given sound-reinforcement scenario will, at a fundamental level, have very comparable “audio circuit” performance to anything else capable of handling that scenario. Modern manufacturing of gear is such that pretty much anything, when run sanely and not engaging in transduction, will have low noise, imperceptible distortion, and transfer response that’s linear from direct-current to dog-whistles.

In other words, there’s no point in looking at SNR, distortion, and frequency response numbers on a mixer’s spec sheet, because it’s all going to be great.

It might not be magic, but it will pass signal in a straight line as long as a component hasn’t failed, and you aren’t hard-clipping the poor thing.

So forget about finding the unit with the best numbers.

Instead, get your mitts on the control surface (whether real or virtual), and figure out if you like how the thing behaves as a tool for intense, realtime munging of loud noises. Does the soft-patching make sense to a rational human? How about to an irrational human on the verge of panic, because something went wrong and the show is 30 seconds from downbeat? Can you make your common routing needs happen without getting lost? If you have preferred EQ setups that you like to use, can you dial them up without struggling? Is it easy to make any built-in compressors and gates act in a way that makes sense? If there are onboard FX engines, can you get the basic delay and reverb sounds you prefer?

These functional considerations are orders of magnitude more important than any subjective sound-quality difference you encounter, especially because they directly affect the “macro-level,” subtle-as-a-kick-in-the-face sound-quality that comes from really messing with an input. At least consider believing me when I say that you don’t actually care about whether or not one console seems to have “slightly deeper and more 3D” bass than another. First, it probably doesn’t – you’re probably just running the “better” console a little louder, or you moved a bit after patching your reference material into the different unit. Second, the tiny little worries evaporate in an instant when the real problem is a musician who “can’t hear the other guitar at all, dude.”

A miniscule difference in distortion characteristics won’t mean squat when the band is 110 dBC continuous in the back of the room without any help from the PA. A 2 dB better noisefloor isn’t worth arguing about when the space is filled with 100 people who are all shouting over each other.

Now…if you’ve got all the basics down, and you’ve found a few different desks that you enjoy using, you’re now ready to nitpick tiny, sonic details. If you’re into that, and you’ve got the time, and the money is all figured out, have at it! If you get a kick out of finding the special mojo, don’t let anyone stop you.

All I’m saying is that the “big mojo” of how comfortable you are with the console as an “audio wrench” matters a lot more. That’s what’s really and immediately going to precipitate what musicians and audio members are going to notice. As is so often true in this business, the ordering of your priorities list is critical.


The Behringer X18

Huge value, especially if you already have a tablet or laptop handy.

Please Remember:

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

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From where I’m standing, the X18 is proof that Behringer should stop fooling around and make a rackmountable X32 with full I/O. Seriously – forget about all the cut-down versions of the main product. Forget about needing an extra stagebox for full input on the rackable units. Just package up a complete complement of 32X16 analog, put a DSP brain inside it, and sell the heck out of it.

I say this because the X18 is a killer piece of equipment. It packages a whole ton of functionality into a small space, and has only minor quirks. If someone without a lot of money came to me and asked what to use as the core of a small-but-mighty SR rig, the XAir X18 would be high on my list of recommendations.

Software Breaks The Barriers

We’ve hit a point in technology where I don’t see any economic reason for small-format analog mixers to exist. I certainly see functionality reasons, because not everybody is ready to dive into the way that surfaceless consoles work, but any monetary argument simply fails to add up. With an X18, $500 (plus a laptop or tablet that you probably already have) gets you some real big-boy features. To wit:

Channel-per-channel dynamics.

Four-band, fully parametric EQ on all inputs and outputs, plus an additional hi-pass filter that sweeps up to 400 Hz.

Up to six monitor mixes from the auxiliaries, each send configurable as pre or post (plus some extra “pick off point” options).

Four stereo FX slots, which can be used with either send-model or insert-model routing as you prefer.

Sixteen, full-blown XLR inputs with individually(!) switchable phantom.

A built-in, honest-to-goodness, bidirectional, multitrack USB interface.

Full console recall with snapshots.

Mute groups (which I find really handy), and DCA groups (which other people probably find handy).

A built-in wireless access point to talk to your interface device.

Folks, nothing in the analog world even comes close to this kind of feature set at this price point. Buying an analog mixer as a backup might be a smart idea. Starting with an analog mixer because all this capability is overwhelming is also (possibly) a good idea. Buying an analog mixer because it’s cheaper, though, is no longer on the table. Now that everything’s software, the console’s frame-size and material cost no longer dictates a restricted feature set.

I’ll also say that I’ve used X32 Edit, which is the remote control software for Behringer’s flagship consoles. I actually like the XAir software slightly better. As I see it, X32 Edit has to closely emulate the control surface of the mixer, which means that it sometimes compromises on what it could do as a virtual surface. The XAir application, on the other hand, doesn’t have any physical surface that it has to mirror, and so it’s somewhat freer to be a “pure form” software controller.

Anyway, if you really want to dive into mixing, and really want to be able to respond to a band’s needs to a high degree, you might as well start with an X18 or something similar.

Ultranet

I didn’t list Ultranet with the other features above, because it exists outside the normal “mixing functionality” feature stack. It’s also not something you can make work in a meaningful way without some significant additional investment. At the same time, Ultranet integration was what really made the X18 perfect for my specific application.

We wanted to get the band (in this case, a worship band for church) on in-ears. In-ears can be something of a convoluted, difficult proposition. Because of the isolation that’s possible with decent earbuds, getting everybody a workable mix can be more involved than what happens with wedges; Along with assuring that monitor bleed can’t hurt you, you also get the side effect that it doesn’t help you either. Further, you still have to run all your auxiliaries back to the IEM inputs, and then – if you’re running wired – you have to get cables out to each set of ears. The whole thing can get tangled and difficult in a big hurry.

The Ultranet support on the X18 can basically fix all that – if you’ve got some extra money.

Paired up with a P16-D distribution module that links to Ultranet-enabled P16-M personal mixers, each musician can get the 16 main input channels delivered directly to their individualized (and immediate) control. If a player needs something in their head, they just select a channel and crank the volume. Nobody else but that musician is affected. There’s no need to get my attention, unless something’s gone wrong. Connections are made with relatively cheap, shielded, Cat6 cables, and the distribution module allows both signal and power to run on those cables.

The “shielded” bit is important, by the way. Lots of extra-cheap Ethernet cables are unshielded, but this is a high-performance data application. The manufacturer’s spec calls for shielded cable, so spend just a few bucks more and get what’s recommended.

Depending on your needs, Ultranet can be a real chunk of practical magic – and it’s already built into the console.

The Quirk

One design choice that’s becoming quite common with digital desks is that of the “user configured” bus. Back in the days of physical components, never did the paths of “mix” and “auxiliary” buses meet, unless you physically patched one into another somehow. Mix buses, also called subgroups, would be accessed via a routing matrix and your channel panner. Aux buses, on the other hand, would live someplace very different: The channel sends section.

In these modern times, it’s becoming quite common for buses to do multi-duty. From a certain standpoint, this makes plenty of sense. Any bus is just a common signal line, and the real difference between a sub-group bus and an aux bus comes down to how the signal gets into the line. When it comes right down to it, the traditional mix sub-group is just a post-fader send where the send gain is always “unity.”

Even, so, may of us (myself included) are not used to having these concepts abstracted in such a way. In my case, I was used to one of two situations: Dedicated buses existing in fixed numbers and having a singular purpose, or to an effectively unlimited number of sends that could be freely configured – but that always behaved like an aux send.

In the case of the X18, the “quirk” is how neither of those two situations is the chosen path. X18 buses exist in fixed numbers, but are not necessarily dedicated and don’t always behave like an aux send. When a bus is configured to behave as a sub-group for certain channels, it is still called a send and located where the other sends are found. However, its send gain is replaced with an “on” button that either allows post-fader, unity-gain signal to flow, or no signal to flow at all. Now that I’m used to this idea, the whole thing makes perfect sense. However, it took me a few minutes to wrap my brain around what was going on, so I figured I ought to mention it.

Other than my minor befuddlement, there’s nothing I don’t like about the X18. It’s not quite as capable as an X32, but it’s not a “My First Mixer” either. It’s actually within shouting distance, features wise, of the more expensive Behringer offerings. There’s a lot of firepower wrapped up in a compact package when it comes to this unit, and like I said, one of these would be a great starting point for a band or small venue that wants to take things seriously.


Thermodynamics, System Coverage, And The Cost Of Lunch

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

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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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.


Just What Signal Is It, Anyway?

This business is all about electricity, but the electricity can mean lots of different things.

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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A fader, an XLR cable, and an Ethernet cable walk into a bar.

None of them could have ducked, because cables and faders can’t walk into a bar anyway. Besides, they don’t play nice with liquids, if we were talking about the other kind of bar.

Look, some jokes just don’t work out, okay?

Every object I mentioned above deals with electricity. In the world of audio it’s pretty much all about electricity, or the sound pressure waves that become (or are generated by) electricity. What trips people up, though, is exactly what all those signals actually are. An assumption that’s very, very easy to make is that all electrical connections in the world of audio are carrying audio.

They aren’t.

The Three Categories

In my experience, you can sort electrical signals in the world of audio into three “species:”

  • Audio signals.
  • Data signals that represent audio.
  • Signals that represent control for an audio-processing device.

Knowing which one you actually have, and where you have it, is critical for understanding how any audio system or subsystem functions. (And you have to have an idea of how they function if you’re going to troubleshoot anything. And you’re going to have to troubleshoot something, sometime.)

In a plain-vanilla audio signal, the electrical voltage corresponds directly to a sonic event’s pressure amplitude. Connect that signal – at an appropriate drive level – to a loudspeaker, and you’ll get an approximation of the original noise. Even if the signal is synthesized, and the voltage was generated without an original, acoustical event, it’s still meant to represent a sound.

Data signals that represent audio are a different creature. The voltage on the connection is meant to be interpreted as some form of abstract data stream. That is to say, numbers. The data stream can NOT be directly converted to audio by running it through an electrical-to-sound-pressure transducer. Instead, the data has to reach an endpoint which converts that “abstract” information into an analog signal. At that point, you have electricity which corresponds to pressure amplitude, but not before.

Signals for control are even further removed. The information in such a signal is used to modify the operating parameters of a sound system, and that’s all it’s good for. It is impossible, at any point, for that control signal to be turned into meaningful audio. The control signal might be analog, or it might be digital, but it never was audio, and never will be.

The Console Problem

Lots of us who louderize various noises started on simple, analog consoles. Those mixers are easy to understand in terms of signal species, because everything the controls work on is audio. Every linear or rotary fader is passing electricity that “is” sound.

Then you move to a digital console.

Are those faders passing audio?

No.

Ah! They’re passing data that represents audio!

Nope.

I have never met a digital mixing desk that does either of those things. With a digital console, the faders and knobs are used for passing control data to the software. With an analog console, the complete death of a fader means the channel dies, because audio signal stops flowing. With a digital console, a truly dead fader doesn’t necessarily stop audio from flowing through the console; It does prevent you from controlling that channel’s level…until you can find an alternate control method. There often is one, by the way.

And then there’s the murky middle ground. More full-featured analog consoles can have things like VCAs. Voltage controlled amplifiers make gain changes to an analog audio signal based upon an analog control signal. A dedicated fader for VCA control doesn’t have audio running through it, whereas a VCA controlled signal path certainly does.

And then, there are digital consoles with DCAs (digitally controlled amplifiers), which are sometimes labeled as VCAs to keep the terminology the same, but no audio-path amplifiers are involved at all. Do your homework, folks.

Something’s Coming In On The Wire

I’ve written before about how you can’t be sure about what signal a cable is carrying just by looking at the cable ends. The quick recap is that a given cable might be carrying all manner of audio signals, and you don’t necessarily know anything about the signal until you actually measure it in some way.

There’s also the whole issue of cables that you think are meant for analog, but are carrying digital signals instead. While it’s not “within spec,” you can use regular microphone cable for AES/ EBU digital audio. A half-decent RCA-to-RCA cable will handle S/PDIF just fine.

Let me further add the wrinkle that “data” cables don’t all carry the same data.

For instance, audio humans are interacting more and more with Ethernet connections. It’s truly brilliant to be able to string a single, affordable, lightweight cable where once you needed a big, heavy, expensive, multicore. So, here’s a question: What’s on that Ethernet cable?

It might be digital audio.

It might be control data.

It might even be both.

For instance, I have a digital console that can be run remotely. A great trick is to put the console on stage, and use the physical device as its own stagebox. Then, off a router, I run a network cable out to FOH. There’s no audio data on that network cable at all. Everything to do with actually performing audio-related operations occurs at the console. All that I’m doing with my laptop and trackball is issuing commands over a network.

It is also possible, however, to buy a digital stagebox for the console. With that configuration, the console goes to FOH while attached to a network cable. Because the console has to do the real heavy-lifting in regards to the sound processing, digital audio has to be flying back and forth on that network connection. At the same time, however, the console has to be able to fire control messages to the stagebox, which has digitally remote-managed preamp gain.

You have to know what you’ve got. If you’re going to successfully deploy and debug an audio system, you have to know what kind of signal you have, and where you have it. It might seem a little convoluted at first, but it all starts to make logical sense if you stop to think about it. The key is to stop and think about it.