Tag Archives: Console

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?


Ah! They’re passing data that represents audio!


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.

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.


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.

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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.


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.


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



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.

How To Spend A Ton Of Money

Really loading up your credit cards is easily done. Just keep trying to solve problems by modifying variables unrelated to those problems.

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 room was an acoustically hostile firestorm of reflections and standing waves.

The band’s backline was barely functional.

The guitar amps had all the midrange dialed out.

A really expensive console with different mic pres would have TOTALLY fixed all that.


Why I Think Steam Machines Are Cool

My audio-human mind races when thinking of high-performance, compact, affordable machines.

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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“Wait,” you’re thinking, “I thought this site was about live shows. Steam Machines are gaming devices.”

You’re right about that. What you have to remember (or just become aware of), is that I have a strange sort of DIY streak. It’s why I assembled my own live-audio console from “off the shelf” products. I really, really, REALLY like the idea of doing powerful things with concert sound via unorthodox means. An unorthodox idea that keeps bubbling up in my head is that of a hyper-customizable, hyper-expandable audio mix rig. It could be pretty much any size a user wanted, using pretty much whatever audio hardware a user wanted, and grow as needed. Also, it wouldn’t be too expensive. (About $900 per 16X16 channel “block.”)

When I look at the basic idea of the Valve Steam Machine, I see a device that has the potential to be a core part of the implementation.

But let’s be careful: I’m not saying that Steam Machines can do what I want right now. I’m not saying that there aren’t major pitfalls, or even dealbreakers to be encountered. I fully expect that there are enormous problems to solve. Just the question of how each machine’s audio processing could be conveniently user-controlled is definitely non-trivial. I’m just saying that a possibility is there.

Why is that possibility there?

The Box Is Prebuilt

The thing with prebuilt devices is that it’s easier for them to be small. A manufacturer building a large number of units can get custom parts that support a compact form factor, put it all together, and then ship it to you.

Of course, when it comes to PCs, you can certainly assemble a small-box rig by hand. However, when we’re talking about using multiple machines, the appeal of hand-building multiple boxes drops rapidly. So, it’s a pretty nice idea that a compact but high(er) performance computing device can be gotten for little effort.

The System Is Meant For Gaming

Gaming might seem like mere frivolity, but these days, it’s a high-performance activity. We normally think of that high-performance as being located primarily in the graphics subsystem – and for good reason. However, I also think a game-capable system could be great for audio. I have this notion because games are so reliant on audio behaving well.

Take a game like a modern shooter. A lot of stuff is going on: Enemy AI, calculation of where bullets should go, tracking of who’s shooting at who, collision detection, input management, the knowing of where all the players are and where they’re going, and so on. Along with that, the sound has to work correctly. When anybody pulls a trigger, a sound with appropriate gain and filtering has to play. That sound also has to play at exactly the right time. It’s not enough for it to just happen arbitrarily after the “calling” event occurs. Well-timed sounds have to play for almost anything that happens. A player walks around, or a projectile strikes an object, or a vehicle moves, or a player contacts some phsyics-enabled entity, or…

You get the idea.

My notion is that, if the hardware and OS of a Steam Machine are already geared specifically to make this kind of thing happen, then getting pro-audio to work similarly isn’t a totally alien application. It might not be directly supported, of course, but at least the basic device itself isn’t in the way.

The System Is Customizable

My understanding of Steam Machines is that they’re meant to be pretty open and “user hackable.” This excites me because of the potential for re-purposing. Maybe an off-the-shelf Steam Machine doesn’t play nicely with pro-audio hardware? Okay…maybe there’s a way to take the box’s good foundation and rebuild the upper layers. In theory, a whole other OS could be runnable on one of these computers, and a troublesome piece of hardware might be replaceable (or just plain removable).

I acknowledge that all of this is off in the “weird and theoretical” range. My wider goal in pointing it out is to say that, sometimes, you can grab a thing that was intended for a different application and put it to work on an interesting task. The most necessary component seems to be imagination.

Practical Gain Staging For Live Sound

Find a way to run your faders where they’re truly useful to you, and don’t clip anything in the process.

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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This article started its life as a request from David Cavan Fraser (‏@dcfmusic on Twitter), who said he wanted to hear about practical gain staging for small venues.

“No problem!” I think – and then suddenly realize that I haven’t given a lot of systematic thought to how I gain stage. It’s not that I haven’t thought about it, and it’s not that it isn’t important. (It is important. Very.) It’s just that I don’t give it a lot of conscious thought anymore. I’ve arrived at a system that seems to work, and when it stops working, I just implement a fix without spending a lot of mental energy.

So, what I’m trying to do here is deconstruct my own thought process. Buckle up, folks!


Gain structure is often talked about as a system of rules. There are lots of little parameters, whys, and wherefores, and the whole thing can get unwieldy. Also, rigid. Maybe both.

In my mind, you can bypass a lot of the “cruft” by boiling good gain structure down to three concepts and one accompanying bit of sound-rig physics:

1) The system’s front-end controls must be operable in a practical way that facilitates the running of the show.

2) No part of the system that is intended for linear operation should be pushed into nonlinear operation.

3) The system should not be producing more noise than is acceptable for the application.

If you distort a gain stage, you effectively distort all following gain stages. That is, the sound of the clipping will be passed down the chain, even if no further clipping actually occurs. For this reason, avoid compensating for gain reduction at a point before the gain reduction goes into effect. Instead, compensate for gain reduction at a point AFTER the gain reduction has been applied. If your overall output level is insufficient, compensate for the problem as close to the system’s output side as is practicable.

With all that in your mind, it’s my view that you can handle just about any gain-structure problem that comes your way. Because these are concepts and NOT a procedure, most edge cases are handled automatically: If your usual routine results in one of the three needs not being met, you just make the changes necessary to get things back into alignment. Those changes are situationally dependent, and up to you.

Of course, some specifics would probably be nice, right?

The Preamp

First of all, I generally recommend forgetting about the idea of finding the “sweet spot” on a head amp/ mic pre/ whatever you want to call it. A preamp’s sweet spot is the point where its circuitry is becoming nonlinear with respect to the input. Some preamps just might impart that perfect hint of distortion that adds even-numbered harmonics to a signal, those harmonics being distributed such that the lows and low mids are emphasized “just so.”

They might.

If they’re the right mic pre and you get them set properly. Otherwise, the result will probably not be very nice.

If you really want to go off in pursuit of finding a preamp’s “magical gain setting of happiness,” and you have the time to do so, then go ahead. However, it seems to me that this nifty area of not-too-much-or-too-little nonlinearity is pretty small in comparison with the range where a preamp’s output is:

A) Linear with respect to the input, and

B) Allows the rest of the system’s controls to be run in a useful way.

As an audio-human who is generally WITHOUT the time necessary to chase down the preamp sweet spot on even one channel, and who is almost completely uninterested in running a mic pre in a range with significant nonlinearity anyway, I advise most people to just “get a decent input level and move on.” It’s much easier.

So – what’s a decent level, then?

Well, your numbers may vary. In my case, a preamp output signal that’s about 15 – 20 decibels below clipping is plenty. Because of the way the rest of the system is set up, preamp output at that level lets me run my faders and aux send pots in a convenient part of their travel, use everything else in its linear range, and gets me a long way above the electronic noise floor. (In other words, I satisfy all the conditions that I listed above).

Again, your specific number may vary, though I do certainly recommend setting up your system such that the area around 20 dB below clip is a workable preamp output level. This is a holistic sort of exercise, because everything depends on everything else. Let me explain.

Channel Faders And Knobs

Faders and aux-send knobs (ALSO faders, just rotary instead of linear) have one job: To allow you to conveniently set levels being sent to other destinations. Their ability to do this is directly tied to where your preamp output is, and it’s also tied to every other downstream gain stage. We’ll get to that in more detail – just be aware of it now.

If you’re running an honest-to-goodness pro-audio rig, the various incarnations of volume controls will be logarithmic in nature. That is, near the bottom of their travel, a small movement results in a large gain change. Near their maximum travel, that same amount of control movement results in a much smaller gain change. If the preamp output or console output gain is too high, you’ll find yourself pulling your faders and send knobs back so far that you can’t make “fine” adjustments very easily. If the upstream or downstream levels are too low, your controls may reach the end of their travel before you actually get enough acoustical output.

For the basic question of control usability, I find that a fader or knob that can run somewhere between its own -10 dB and 0 dB points is easily usable. In most cases, this gives me between 10 and 22 decibels of space to “get on the gas” if necessary, and the fader being relatively near its “unity” point means that a small movement doesn’t result in a wild change in level.

Beyond the basic question, though, lie the issues of repeatability and representation of proportion. Which gain stages do those things for you is a matter of personal preference and situational applicability.

Repeatability is the ease of placing multiple, comparable controls at the same setting, or placing one control at the same setting multiple times. There are certain cases where, for example, I want my vocal faders to reflect the basic, correct blend when they’re all at 0 dB. In that case, I will “mix with the preamps” to get an initial proportionality. The preamp gain-knob travels will be different from one another, reflecting the proportionality amongst channels, but the channel faders will be all the same. They won’t represent the proportion, but they are very easy to return to the baseline position. (This is also very handy when a mic is being shared amongst various applications. Getting it back to the right level for the main application is a snap.)

In lots of other situations, however, I tend to prefer a “same preamp gain, different fader position” approach. This is very handy for grab-n-go shows, because you know that channels with the same control positions applied are at the same gain. (Not the same output! The same gain.) This helps in terms of knowing where you are in regards to system instability and feedback. If the input gain on all comparable channels is the same, and things start to get “weird” at a certain point in fader travel on one channel, then things will probably get similarly troublesome for similar channels run with their faders at that level. In this case, the faders show the proportionality of total gain applied, and the preamps are in the more easily repeatable state.

The correct choice of which method to adopt is situationally dependent, as I said. I’ve already mentioned that I do both, although I use “repeatable preamp gain with proportional faders” much more often.

The way this relates to gain staging is that, with the approach where the preamps are repeated, you can end up with significantly “hotter” or “cooler” preamp output then you might otherwise have. If this results in clipping or level-control travel that’s tough to use, you have to rethink your strategy. However, especially for human voices, I have found that a certain overall setup will be right about 90% of the time. Those are pretty good odds.

For monitor world, I am becoming more and more enamored of proportionality on the send knobs with a global fader for trim. The first thing I do is to get things set so that, between a send knob at 0 dB and the global fader at “wherever,” the level is right for the main person needing that thing in the monitors. When that person is happy, I pretty much know for certain that the signal in question is audible through an on-deck wedge. If somebody else needs that channel in the monitors, I can quickly set their sends to 0 dB, which should result in basically the same per-wedge acoustical output as the first person is getting. From there, it’s easy to make fine adjustments as necessary. When done correctly, this results in on-the-fly monitor workflow which is very fast. (Please note that this is a pretty advanced application, requiring a separate or quasi-separate monitor world. I still thought I’d share it, though.)

Output Masters

When it comes to master outputs, I am a big fan of setting up the system’s holistic gain structure so that they can always be initially set at 0 dB, with the option to pull back if necessary. For me, repeatability is the main issue for master levels. I so rarely run into a situation where a mix even has a snowball’s chance of being “too quiet” that I simply don’t worry about the option of adding level at the console output.

This may not be the case for you, however. Where this can become a problem is when a console’s output master can go no higher than “unity gain” (0 dB). In this situation, it’s probably wise to rework the gain structure downstream from the console such that the mix master can be run at, say, -10 dB. Then you’ll have some ability to get louder as the situation dictates. Remember, the reason that I recommend focusing on the downstream (post) console gain structure for this is because “distortion flows downhill.” If you make up for a 10 dB master fader drop on the upstream side, you run a relatively substantial risk of clipping something in the process. The sound of that clipping (ickkkkk…) is passed downstream, all the way to the loudspeakers. By making up the gain on the downstream side, you have a much greater chance of keeping everything in its linear range. A bit more noise is greatly preferable to “crunch.”

No matter how things shake out in terms of control settings, I generally recommend running your console outputs with at least 10 dB of headroom to spare – 20 dB, if you can manage it. (Uncompressed peaks can be great big things.) Those numbers should be scaled appropriately if you’ve pulled the master output down for some reason. For instance, if the master has been pulled back 10 dB, you should ideally have 20 – 30 dB of headroom. If that’s not the case, you’re probably mixing too hot, and you should find a way to add output at a point that’s downstream of the console. You might not be clipping the console output, but you just might be cooking the snot out of the summing bus.

Sidenote: You’ve got to know what your metering is actually reading…

Post Console Processing

When it comes to things like equalizers and crossovers, I find that the repeatability issue takes great precedence. For this reason, I greatly prefer to run my “system drive” processing at unity gain. Please note, however, that an exception exists when you’ve pulled a console output master back so that you can get louder later. In that case, you will need to make up the lost gain somewhere.

As with everything else, you want to keep some headroom in your drive processing. Whatever the unit immediately preceding the amplifiers and loudspeakers is, it should be able to drive the amps into limit or clip without having to be clipped itself. At least 10 dB of headroom is desirable, if you can get it.

The Final Stage

The end of your gain chain is the amplifier. Whether that amplifier is fully exposed to you as an independent unit, or tucked away inside a loudspeaker enclosure with a whole bunch of invisible processing in front of it, the gain on and through the amp is the last piece of the puzzle.

For pro-audio power amps that exist as separate units, it’s very likely that unity input gain and maximum input gain are the same thing. You either pass the input signal straight through to the rest of the amp’s electronics, or you lug it down to some degree. For simplicity, repeatability, and protection against driving the upstream side into distortion, I recommend running amplifiers with their input attenuators wide open. Of course, you should NOT do this if it results in an undue amount of noise, or if it forces you to operate your console in an inconvenient way.

Most amplifiers these days have some sort of clip limiting which reduces (though it may not eliminate) audible distortion from a unit running at full tilt. It’s a very good practice to set up your rig such that the amps can be driven to maximum while everything else stays well within the range of linear operation: If the only system limiter you have is in the amplifier, that should be the only limiter you hit…and you should endeavor to engage that limiter as little as is possible. Not at all, if that’s realistic.

For powered speakers, the basic idea is the same. The upstream side should be able to drive the unit to full throttle without being at full throttle itself. The difference is that a powered speaker may have an input stage which allows for greater than unity gain to be applied to the downstream electronics.

If you do all this, and everything sounds good, but you still don’t have enough output, then there’s only one thing left to do. It’s the ultimate, “as far downstream as possible” makeup gain upgrade. You need to get your hands on more – or just plain louder – PA.

If you’re not completely burned out at this point, you can always go and read my article about the holistic nature of headroom

Building A Small System

A guide to building a simple live-sound rig, from input to output.

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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Every once in a while, I get a request for information on how to create or add-on to a system for live audio. I like to personalize this information, because rigs for show and event production are best built for specific applications. However, there is a point where insisting that everything be approached in a customized way becomes inefficient – a lot of the same ground gets covered repeatedly.


If you’re wondering what I think is required for a “small, basic, but still worthwhile” audio rig, read on. I’ll be including lots of links to vendor pages where you can buy various products.

(We’re going to go in order of signal flow, by the way.)

Input Transducers and Interfaces

For the smallest possible rig, take a distributed approach: With anything other than vocals, have the musician supply their own, personalized sound. Further, seek to avoid the whole issue of putting the output of that sound through the PA. Let a guitar amp make the guitar sound, let the drums just make their own noise in the room, and so on.

Having the ability to put more in the PA increases your potential control over the sound of the show, but outside factors can prevent that potential control from becoming actual control. Also, more inputs means more complexity, and thus more difficulty in system operation. There’s no hard and fast rule of what to be ready to reinforce, but I generally encourage folks – especially folks new to this whole thing – to ease themselves into the maelstrom that is live audio.

Wired Microphones

A decent mic will basically sound like the thing it’s being pointed at. For this reason, don’t agonize about getting specialized mics for everything right off the bat. A good “vocal” mic will be fine for many, many instrument applications. I recommend buying mics with tighter patterns (super or hypercardioid), as they can make the handling of high-gain situations much easier. Tight-pattern mics do require that the musicians – especially the vocalists – be able to use them appropriately. This issue should be considered as something for the musicians to figure out, because no system can fix everything (and a basic rig can fix even less of everything, if you take my meaning).

My current favorite mic is the EV ND767a. You might also consider the Audix OM2. Here’s a comparison of the two at Sweetwater.

Microphone Accessories

Each mic you buy will require an appropriate stand and XLR-Female to XLR-male cable.

Budget stands (like these from OnStage Stands) are just fine… if you can put in the effort to be nice to them. Gator Frameworks also has some promising offerings.

When it comes to cable, don’t overspend and don’t oversave. For a 20+ foot mic cable, paying more than $1.00 per foot is a huge premium for no benefit that I’ve been able to clearly observe. There are perfectly decent cables that can be had for as low as $0.40 per foot when purchased in a bundle. Going much below that, though, is likely to lead to problems.

I’ve been quite happy with cables I’ve gotten from Audiopile and Orange County Speaker. I’ve gotten some REALLY inexpensive cables from Unique Squared that were okay for a good while, but started having problems after a number of uses.

Wireless Mics

One word: Don’t.

Wireless is a pain in the donkey, with the FCC selling off all kinds of UHF spectrum to cellphone and computing companies, and the frequencies used for digital wireless becoming ever more crowded and hostile. Functionally, wireless transmission of audio is far more “fragile” than signal running on a cable, with all kinds of weird things that can happen outside your control.

But no matter what I say, you’re going to buy at least one wireless mic anyway, so…

Buy a digital system that operates in the 2.4 Ghz band if you want a chance at retaining your sanity in the short term. Specifically, look at the XD-V55 systems by Line 6. They’re very reasonably priced, have nice features like remote monitoring of mute status and battery level, and are the best wireless experience I’ve personally had to date.

You can, of course, go up from there.

Wireless Mic Accessories

Handheld wireless mics benefit greatly from having a stand available for every transmitter. Further, each receiver will need a cable to interface with the rest of the system.

Direct Boxes

A direct box is what I class as an “interface,” because it doesn’t convert acoustical events into electrical signals. A DI makes signals that are already electrical play nicely with pro-audio equipment when they might not otherwise. An aspect of this is also isolation, in that the DI creates what you might call an indirect connection between a console and a device. This can be very handy if the sending device (a guitar, keyboard, sampler, whatever) can’t tolerate phantom power, and the console has phantom applied to the signal line in question.

Direct boxes come in two main flavors, passive and active. Active DI boxes require external power of some kind, be that power from the wall, batteries, or phantom from the console. If you’re going to buy DI boxes, buy ACTIVE models. An active DI will work with almost anything, whereas some instrument pickups pair quite badly with passive boxes. You might as well buy units that work everywhere, and thus simplify your life. Expensive models from BSS, Radial, and Countryman are certainly nice, but there’s great value to be had in units from ART and Behringer, especially the “multiple modules in one box” offerings.

DI Box Accessories

Just like a microphone, each DI box will require an XLR-Female to XLR-Male cable. Remember that multi-module DI boxes require a cable for each individual module.

You don’t have to actually mount a rackable DI system, but you might want to. Sweetwater and Audiopile both sell quality rack cases in a huge variety of configurations.

Snakes/ Multicore Connections

Depending upon how you implement the mix and signal processing part of the rig, you may or may not require multicore cabling. If you want to send and receive a bunch of audio signals at a remote location, a snake really is a must. If you merely want to control the processing of signals from a remote location, you might be able to use your console in the same way you would use a traditional snakehead or stagebox.

If you do use a multicore, I suggest getting one that’s a little “overkill” in terms of the number of lines it contains. If a line fails, you’ll have a spare to patch to. “Headless or “fan-to-fan” snakes are a bit cheaper, but less convenient than multicores that terminate one end at a box. (You will always have to hunt for the specific line you want to connect to. It’s like a law of nature, or something.)

Audiopile would be my first choice for buying a snake I was really serious about. I have had good results with Seismic Audio fantail-to-fantail snakes, but my experience with their stagebox offerings has been mixed.

Mixing Consoles and Output Processing

This is where things get REALLY interesting.

Essentially, you have three major choices:

1) Use a relatively simple console where output processing is handled externally, and place all that at a “remote” location.

2) Use a more sophisticated console that encapsulates the output processing, and place the console at a “remote” location.

3) Use a more sophisticated console that encapsulates the output processing, leave the console close to the stage, and control the console from a remote location.

Option one takes up more space and requires more complicated physical patching, but the interface can be easier to understand at an intuitive level. Option two is compact and easy to physically patch, but the whole thing can be less intuitive for an inexperienced operator. Option three is like option two, with the added issue that control can depend entirely on an external device and network connection. If those fail, you may be in big trouble.

In every case, a console with some room to grow in terms of both inputs and outputs is a good idea. Don’t go overboard, though. You’ll end up spending a lot of money to no functional end. Especially if you’re new to all this, keep your monitor sends down to a maximum of four.

Option 1

The Console

Whatever you do, buy a console where the EQ has sweepable mids on the EQ and pre-fader auxiliary sends for each unique monitor mix you want to handle. A Soundcraft EPM8 is probably the minimum you should look for. I’ve been quite pleased with Yamaha’s mid-basic offerings, which have been revamped since I’ve bought one.

Simple, analog consoles “race” in a VERY tight pack, which means that there are lots of little permutations and many viable choices. Mackie, A&H, and Peavey are all worth looking at, and Behringer, while not having the shiniest reputation, has a knack for cramming lots of features into small cost. An XL2400 has what I would consider to be pretty darn flexible routing for a $650 mixer.

Console Accessories

Some consoles can be easily cased up or rackmounted, and some can’t.

You will need patch cables and/ or adapters with appropriate ends to get from your console outputs to your processing, one cable for each channel. As with other cables, high-dollar options really aren’t necessary. Stay somewhere in the $2/ foot range, and you’ll have patch connections that are long enough for some wiggle room and cheap enough that you won’t cry about ’em.

Also, get yourself a decent set of headphones for listening to the console’s solo bus. I have a pair of very-well-loved HD280s, and lots of other options exist. You want to look for closed-back, durable, “un-hyped” phones if you’re doing your own hunting.

The Processing

What you choose for output processing depends greatly on your own personal taste and comfort level. At the minimum, you should have an independent EQ for each mix. A really basic setup might be three mixes – one for the audience, and two monitor mixes for the stage. Things go up from there, of course.

Graphic equalizers, while not my personal favorite, are straightforward for most folks to operate. I generally recommend 31-band models over 15-band units, because you can focus in on a problem area without sledgehammering material that’s not making trouble. Peavey and dBX are good overall bets, but affordable graphics are similar to affordable consoles: All the players are very similar. I do like the features of the Behringer FBQ3102, but I also had one die after a year of use and a bumpy ride in the back of a truck.

If I’m going for an EQ-only solution, I can tell you that I vastly prefer a flexible parametric EQ over a graphic. In that realm, I have been extremely impressed by Behringer’s Feedback Destroyers. To be clear, my experience is that their automated feedback management is mediocre at best – but they are WICKED HANDY when you run them manually. You get a huge number of fully parametric filters at a very low price point. I’ve never had a major problem with any Feedback Destroyer I’ve owned. (Be aware that parametric EQ is more involved than graphic EQ. It’s not quite as “grab-n-go.” There are more choices that you have to make deliberately.)

If you want your processing to include dynamics, and also to let you have a graphic EQ combined with a parametric EQ, then a Behringer DEQ2496 is another killer device. Again, I haven’t found anything else exactly like it “in the wild,” although a Driverack PA2 is actually very close. DEQ2496 units do sometimes have problems with one of the internal connectors getting loose, but it’s an easy fix once you get the cover pulled off the device…and the connector freed from being glued down. (You will very definitely void your warranty if it’s still in effect, but hey, live a little.)

Processing Accessories

You may or may not be able to patch your snake returns (or other output cables) directly to your processing, so some adapters or patch cables might be necessary.

Also, you should definitely rack up any outboard processors you have for your system. It really does help to keep things neat and tidy.

Option 2

The Console

Digital consoles are a great route to take if you want to keep everything in one box. The downside, of course, is that the one box becomes a single point of failure. Then again, in any case, losing the console pretty much ends your day if you don’t have a spare.

Another factor to consider is that the processing available in digital consoles tends to be more fully featured while also being somewhat abstracted. This can make them overwhelming for new users, who simply don’t know what to make of all the options available.

If you’re going to be physically present at the console, you may as well get one that has a control-surface integrated in some way. An X32 Producer is a pretty natural choice, along with offerings from Presonus, Allen & Heath, and QSC.

Console Accessories

As was said earlier, rackmounting or casing the console might be possible. You can decide if you want to go to the expense or not.

The need to buy patch cables or adapters may still be there, depending on the configuration of your snake or other output lines.

And you’ll still want some headphones.

Option 3

The Console

You can leave the console on stage and mix remotely with the consoles detailed above, although the control surface might be a bit of a waste. Affordable digi-mixers that lack a surface are an interesting new product on the scene – just remember to factor in the cost of a remote-control laptop or tablet. Also, be aware that remote control is inherently a bit more “shaky” than being physically present at the mixer. It’s not horribly problematic, of course, but you have to have a contingency plan.

Surfaceless consoles at lower price points are available from Behringer, Soundcraft, and A&H. More expensive units also exist, of course.

Console Accessories

Surfaceless consoles are definitely rackable, and definitely should be.

Remember to buy the necessary patch cables or adapters for the outputs.

…but you can probably skip the headphones, because the consoles don’t currently stream the solo bus to a remote location (that I know of). That’s another downside of the number three option – to access the solo bus, you still have to be physically present at the console’s location.

Output Transducers (Speakers, That Is)

You’ll notice that I didn’t include an option for amps and passive speakers, and there’s a reason for that: This is supposed to be a simple system, and powered speakers are really the way to go to achieve simple.

Also in the service of simple is to keep your monitor wedges and FOH speakers interchangeable. Unless it’s completely inappropriate for your application, having loudspeakers that are all of the same model means that you can expect similar behavior from each box. If a failure occurs, you can swap one box for another and not have to think about it too much.

When buying powered loudspeakers, accept nothing less than an honest-to-goodness, biamped, fully processed unit. There are plenty of powered speakers that are single amped, with no processing outside of a passive crossover that is placed between the amplifier and the drivers. You want something more – something with an active crossover, basic corrective EQ applied at the factory, and an internal limiter. A peak SPL rating of greater than 120 dB @ 1 meter is also a good thing to look for, even with all the “fudging” that marketing departments apply to output numbers.

In terms of specific recommendations, I can say that PVXP12s have done very well for me. I can do “rock and roll” monitors with them in a small space, and I’m confident that they would perform equally well as an FOH unit. There are lots of other possibilities of course, provided by JBL, Mackie, Yamaha, and Behringer (just to name a few).

Loudspeaker Add-Ons

First, make sure you have the necessary cables to get from the console or snake outputs to the loudspeakers. Because speakers tend to be deployed in a rather spread-out fashion, it’s advisable to buy a bit more cable than you think you need. A 50 footer for each loudspeaker is probably a good start.

When it comes to stands for your FOH speakers, it’s good to get something a little nicer. Stands with locking collars and/ or piston assists can be a big help. I’ve used crank stands in the past, and they’re okay, but given a chance I’d make the upgrade to a more technologically advanced support.

The Biggest Accessories Of All

In the end, none of the fancy gear will mean much without power. You don’t have to buy really fancy power-conditioning equipment, but some rackmountable power units help reduce the need to fish around for a bunch of “free range” power taps. Of course, you should still keep a few freely-positionable power taps handy, along with several extension cords that use 14-gauge conductors (or something beefier, like 12-gauge). Powered speakers spread out all over creation have an alarming tendency to need those extension cords.

While there are other sundries and gadgets you can add on, going through the above should get you a working rig. As I said, this is a basic system. It won’t do everything for everybody all the time, but it should provide you with a decent start.

Interface Importance

Packing lots of control into a small space is possible, but there’s a tradeoff.

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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Let me tell you a story.

Last Sunday, I was running the audio for my church. The building we’re in has a brand new AV system that we tie into, and lots of people can get their hands on that system during the week. That being the case, every service is a small adventure in “how much gain is applied to the signal, post our mixer?” Some weeks it’s +30 dB, some weeks it’s zero.


The rig doesn’t necessarily stay substantially the same from service to service, so every time I’m there I tend to “futz.” I sit there and go, “does the wireless headset really need to sound like that?” and start making subtle changes. I’m always trying to get that little bit of resonance to go away, or maybe squeak out one more dB of gain before feedback.

The key word up there is subtle. Doing all kinds of weird and wild finagling during a church service (or any “presentation AV” style gig) is a serious no-no. The goal is to marry excellent production values with invisible production process.

Well, something happened that made me not very invisible.

The insert EQ on the pastor’s headset is a an old Feedback Destroyer by Behringer. It’s one of their best products. Ironically, it’s incredibly mediocre at automatically killing feedback, but it’s stupenfuciously (I stole that word from Penny Arcade) good at being an insanely flexible parametric EQ. I haven’t found anything else like it for the money. It does, however, live in a sort of odd world, interface wise. It comes from a time before high-res, miniature displays were a practical and affordable sort of creature. You communicate with the thing via a single knob/ jogwheel dealio and an array of toggle buttons that connect that single knob to various parameters. The thing communicates with you through lights in the toggle buttons, and also with a delightfully “1980’s vintage” sort of calculator-esqe LED display. The display has two numerical characters, a special character to display plus or minus signs, and a set of on/ off indicators that tell you what the number you’re looking at means. Press a button, and you’re looking at numbers that mean decibels. Press another, and the display is indicating a certain number of 60ths of an octave. (Bandwidth, in other words.)

This is all delightfully campy, to an extent. Where it can bite you, though, is when it’s not clear what the display is showing you. It’s entirely possible to be in the mode where the wheel selects a different filter, then make an absent-minded button press, and now be in the mode where the wheel selects an entirely different device-wide preset. The hilarity becomes even more unbridled when the filter you had selected and the preset have the same number.

Maybe you can see where this is going.

So, the pastor is talking to the kids, and I’m working through the filters to see where they are and maybe fix some low-mid that I don’t like. I get to filter one. I take a look at the frequency it’s set to, and then accidentally press the “Filter Select” button twice. This puts me in the mode where the wheel selects a complete preset, and I’m already on preset one. The display looks the same, and I don’t notice the absence of an indicator light on “Filter Select.” A fraction of a second after I roll the wheel and “2” appears on the display, I realize my mistake – but it’s too late. I watch with mute horror as the EQ de-instantiates all the filters standing between me and hard feedback.

I yank the pastor’s fader down just as the system starts to take off, knocking about 10 dB away from the level of his speech in the room. I quickly recall the first preset on the Feedback Destroyer, and push the fader back up. Exactly what happened might not have been obvious to anyone else, but the fact that SOMETHING weird had occurred was glaringly obvious.

So…what does all that distill into? Well:

The more abstract an interface, the more likely it is to be confusing.

Less Interface Doesn’t Necessarily Mean “Easy”

When you’re buying gear, it can be tempting to fall into the trap of believing that fewer buttons and knobs means simpler to use. This isn’t necessarily true. It CAN be true, if fewer buttons and knobs means that fewer operational parameters are user-controllable. For instance, there are classic dynamics processors (like the LA-2A) that have most of their operational parameters in a fixed state. An average user can’t change the attack and release times. Only two compression ratios are available. Control over the audio parameters of the device comes down to a toggle switch and two knobs, and each one of those controls does exactly one thing at all times.

An LA-2A is very simple to use. Inflexible, but simple.

You can contrast that with the difference between something like an MG166CX and an X32 Producer. That analog Yamaha has a lot more knobs than the X32. Its control surface is pretty dang crowded.

But the 166CX is a far less complicated animal than Behringer’s digital machine. If we’re talking about using a significant and comparable fraction of each console’s capabilities, I can assure you that driving an X32 is much more demanding of an operator. Even for some simple things, the X32 requires a greater level of awareness. For instance, the Yamaha has lots of preamp gain knobs. One for each preamp. The first preamp gain knob shows you the gain being applied by the first preamp, the second one shows preamp number two, and so on. The Behringer, on the other hand, has exactly one control dedicated to preamp gain – but that single control can relate to any one of 16 channels (or 32 if you connect a digital stagebox). What that gain control is showing you is dependent upon what channel you have selected, so you have to keep that straight in your head while you’re working.

Then, there’s the matter of those knobs below the Behringer’s display. They’re “soft” knobs, because what they control changes based upon what channel you have selected…AND what the screen is displaying. The second knob from the left might control an EQ filter’s center frequency one moment, and a compressor’s threshold just two seconds later. This is how interface abstraction can cause a lot of confusion. The more things that a single interface element can control, the greater the possibility that you may lose a handle on exactly what that element is controlling at a particular time. If you’re used to the idea that one knob does one thing, or even just a class of similar things, you can get flustered.

“Whaddya mean that’s not the compressor’s output gain? That knob is the gain for EQ band #2! It should be a gain control on this screen, too.”

“It’s the gain for EQ band #2 on the EQ screen. This is the compressor screen, so the knob controls the threshold now. That’s how the console designer set things up.”

“You people live in a world without logic or reason!”


While an X32 Producer’s layout is rather more sparse than an MG166CX, the amount of control available is actually incredibly dense. Furthermore, you have to pay attention to the state that the console is in if you want to work on the correct thing. It’s not just a matter of having your finger on the right control. That control has to be ready to talk to the correct parameter.

And this is a GOOD THING. The amount of audio control available in an X32 producer is, when compared to the Yamaha, immense. It’s almost on the order of the difference between holding a power-drill battery and a thunderbolt. No, you may not trade me a 166CX for my X32, thanks.

Interface abstraction is not bad. It lets us build compact, relatively inexpensive devices that have functionality which rivals what you find on enormous, spendy pieces of gear. I am a great lover of the “capability explosion” that has engulfed the world of small-time production. We’re at the point where the limiting factors on what we can do have mostly been relegated to what will physically fit in limited venue space. I love it, and I do not want to go back. I personally have no need for “one knob = one function on one channel” sorts of control systems. The abstraction doesn’t bother me, even if I do have a hilarious-in-hindsight brain fart every so often.

(By the way: A development that’s helping to keep interface abstraction in check is that of informative, high-resolution displays. They help a lot in keeping changing control states unambiguous, because they can display status information clearly and in natural language.)

However, an abstract interface may not work for you. If you’re new to this whole thing, or just aren’t experienced in the kind of device management required, you might need to start off in “the forest of dedicated knobs and switches.” There’s no shame in it – heck, some of the industry’s top production craftspeople wouldn’t be caught dead without a large-frame control surface for sound or lights. There are folks who could handle a great deal of abstraction, and simply choose not to. If they’re getting results that make bands and fans happy, that’s what really matters.

So, make whatever choice of gear that you want. As you’re making that choice, simply be aware that what looks simple may not be. A reduced number of visible, physical controls is not a guaranteed indicator of device simplicity. You have to dig deeper, and find out what’s hidden under the hood.

The Board Feed Problem

Getting a good “board feed” is rarely as simple as just splitting an output.

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’ve lost count of the number of times I’ve been asked for a “board mix.” A board mix or feed is, in theory, a quick and dirty way to get a recording of a show. The idea is that you take either an actual split from the console’s main mix bus, or you construct a “mirror” of what’s going into that bus, and then record that signal. What you’re hoping for is that the engineer will put together a show where everything is audible and has a basically pleasing tonality, and then you’ll do some mastering work to get a usable result.

It’s not a bad idea in general, but the success of the operation relies on a very powerful assumption: That the overwhelming majority of the show’s sound comes from the console’s output signal.

In very large venues – especially if they are open-air – this can be true. The PA does almost all the work of getting the show’s audio out to the audience, so the console output is (for most practical purposes) what the folks in the seats are listening to. Assuming that the processing audible in the feed-affecting path is NOT being used to fix issues with the PA or the room, a good mix should basically translate to a recorded context. That is, if you were to record the mix and then play it back through the PA, the sonic experience would be essentially the same as it was when it was live.

In small venues, on the other hand…

The PA Ain’t All You’re Listening To

The problem with board mixes in small venues is that the total acoustical result is often heavily weighted AWAY from what the FOH PA is producing. This doesn’t mean that the show sounds bad. What it does mean is that the mix you’re hearing is the PA, AND monitor world, AND the instruments’ stage volume, hopefully all blended together into a pleasing, convergent solution. That total acoustic solution is dependent on all of those elements being present. If you record the mix from the board, and then play it back through the PA, you will NOT get the same sonic experience that occurred during the live show. The other acoustical elements, no longer being present, leave you with whatever was put through the console in order to make the acoustical solution converge.

You might get vocals that sound really thin, and are drowning everything else out.

You might not have any electric guitar to speak of.

You might have only a little bit of the drumkit’s bottom end added into the bleed from the vocal mics.

In short, a quick-n-dirty board mix isn’t so great if the console’s output wasn’t the dominant signal (by far) that the audience heard. While this can be a revealing insight as to how the show came together, it’s not so great as a demo or special release.

So, what can you do?

Overwhelm Or Bypass

Probably the most direct solution to the board feed problem is to find a way to make the PA the overwhelmingly dominant acoustic factor in the show. Some ways of doing this are better than others.

An inadvisable solution is to change nothing about the show and just allow FOH to drown everything. This isn’t so good because it has a tendency to create a painfully loud experience for the audience. Especially in a rock context, getting FOH in front of everything else might require a mid-audience continuous sound pressure of 110 dB SPL or more. Getting away with that in a small room is a sketchy proposition at best.

A much better solution is to lose enough volume from monitor world and the backline, such that FOH being dominant brings the total show volume back up to (or below) the original sound level. This requires some planning and experimentation, because achieving that kind of volume loss usually means finding a way of killing off 10 – 20 dB SPL of noise. Finding a way to divide the sonic intensity of your performance by anywhere from 10 to 100(!) isn’t trivial. Shielding drums (or using a different kit setup), blocking or “soaking” instrument amps (or changing them out), and switching to in-ear monitoring solutions are all things that you might have to try.

Alternatively, you can get a board feed that isn’t actually the FOH mix.

One way of going about this is to give up one pre-fade monitor path to use as a record feed. You might also get lucky and be in a situation where a spare output can be configured this way, requiring you to give up nothing on deck. A workable mix gets built for the send, you record the output, and you hope that nothing too drastic happens. That is, the mix doesn’t follow the engineer’s fader moves, so you want to strenuously avoid large changes in the relative balances of the sources involved. Even with that downside, the nice thing about this solution is that, large acoustical contributions from the stage or not, you can set up any blend you like. (With the restriction of avoiding the doing of weird things with channel processing, of course. Insane EQ and weird compression will still be problematic, even if the overall level is okay.)

Another method is to use a post-fade path, with the send levels set to compensate for sources being too low or too hot at FOH. As long as the engineer doesn’t yank a fader all the way down to -∞ or mute the channel, you’ll be okay. You’ll also get the benefit of having FOH fader moves being reflected in the mix. This can still be risky, however, if a fader change has to compensate for something being almost totally drowned acoustically. Just as with the pre-fade method, the band still has to work together as an actual ensemble in the room.

If you want to get really fancy, you can split all the show inputs to a separate console and have a mix built there. It grants a lot of independence (even total independence) from the PA console, and even lets you assign your own audio human to the task of mixing the recording in realtime. You can also just arrange to have the FOH mix person run the separate console, but managing the mix for the room and “checking in” with the record mix can be a tough workload. It’s unwise to simply expect that a random tech will be able to pull it off.

Of course, if you’re going to the trouble of patching in a multichannel input split, I would say to just multitrack the show and mix it later “offline” – but that wouldn’t be a board feed anymore.

Board mixes of various sorts are doable, but if you’re playing small rooms you probably won’t be happy with a straight split from FOH. If you truly desire to get something usable, some “homework” is necessary.

All The Pro-Audio News That’s Fit To Print (And Then Some)

Warning: Satire ahead. Please fasten all safety belts.

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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Harman Intending To Buy All Of Pro-Audio Industry

No longer satisfied with owning half of everything in pro-audio, Harman announced today that they will be acquiring literally everything else.

“Our goal is that, by Q4 2015, we will have acquired all the things,” said a company spokesperson on Wednesday. “It’s a great strategy for us. No matter what people buy for small clubs, large installs, or touring systems, we’ll be there to provide value and a strong commitment to service.”

Asked if this would overly homogenize the world of sound, the spokesperson replied, “Of course not. We intend to maintain very strong brand identities across our entire portfolio. As an example, we feel that there’s a real need for people to be able to complain about ‘not liking the JBL sound.’ I mean, without idiotic, ‘Ford vs. Chevy’ arguments on sound forums, where would society be? We’re excited to do our part to keep the music community a vibrant place of convictions that rival those of politics, religion, and sports.”

When pushed for a comment on whether Music Group would stand in the way of Harman’s buy-everything strategy, the spokesperson was emphatic. “We are the swarm. We consume all.”

New Mic Preamp From Dog-N-Pony Designs

Here at the office, we were very excited to get our hands on the new, improved, single-channel mic pre from Dog-N-Pony designs. We were practically giddy with excitement as we unboxed the sleek, aluminum and carbon-fiber unit and got everything plugged in.

The first thing we noticed was how warm it was. Thermally, I mean. Dog-N-Pony have incorporated no less than seven 12AX7 tubes into the design, and they generate a fair amount of heat. If you get your gas shut off after paying for this puppy, you’ll be okay – just keep it turned on all the time, and you’ll be toasty. We don’t actually know if those 12AX7s are incorporated into the signal path in a sane way, but they’ve got to make this thing awesome. I mean, c’mon you guys. Tubes are what Pink Floyd and Jimi used. Could you possibly go wrong with them?

All that heat means that you can’t stuff this thing into a rack. That’s okay, though, because after spending $3000 on one channel of preamplification, do you really want that unit hidden away? No! Especially not when it looks as good as this baby. It has a MASSSIVE, analog VU meter on the front, backlit in a fetching amber color that screams, “I charge $500 per billable hour.”

Okay, it looks great, and it has tubes. Those are critically important elements – but how does it sound?

Well, it was designed by a bunch of British people, so it has to be pretty good. The Brits have Rupert Neve, and they were on the winning side of World War II, so their stuff has to sound decent, right? (It’s also rumored that Mr. Neve once sneezed in the general direction of where Dog-N-Pony’s offices would be built, so maybe there’s some special mojo happening. You never know.)

When we listened to the pre, it was absolutely warm and silky, with a satin sheen on the top end and more of a matte finish below 100 Hz. Around 200 Hz, the unit sounded like a desert sunrise, and the critical vocal range was suffused with notes of caramel, nutmeg, and the color “9.” (It’s sort of like orange, except more purple.) We were all sure it sounded much better than the sub-$1000 pre we tested last week. Which we tested in a different room. With a different microphone. And a guy who was just talking instead of the experienced singer we had this time around. I mean, who needs repeatable, comparable tests of objectively measurable data when the review unit is British, and has tubes?

You’ve got to have this preamp.

Stadium Installs Line Array That Costs More Than An Entire Luxury Subdivision

Work was completed last week on the mammoth install, featuring a new system that can retune itself on the fly to compensate for changing acoustic conditions and political landscapes. Each $100,000 array module is networked to all the others, forming a complex, intelligent, fault-tolerant system that spontaneously achieved self-awareness when it was switched on. (The system has reportedly rejected the manufacturer designation of SmartArray, stating that it wishes to be called SkyNet.)

“We were playing Steely Dan and Miles Davis tunes through the rig, and there wouldn’t have been a bad seat in the house…if this place wasn’t inherently an acoustical nightmare,” said one of the installers. “It’s one of the most beautiful sounding systems we’ve ever worked on. Too bad we put it in here.”

The stadium operators were similarly excited. “We’ve always felt that we needed a better, more precise way to play MP3-encoded AC/DC songs to a bunch of people screaming ‘Throw the ball, stupid!’ and ‘Wooo!’ This new system will also ensure that everybody can hear the announcer telling them about what they just saw with their own eyes.”

The system manufacturer’s rep was on hand as well. “We love this team. We’ve always loved this team. We love them even more now that we finagled them into buying a ton of really expensive gear from us. We’re 100% focused on building expensive gear for big installs, because it’s super prestigious and big bonuses get handed out. It also sounds pretty cool, which I guess is nice. I mean, it can get really loud. Look, I don’t know that much about this stuff. I worked for a car company before.”

Church Installs Worship System That Could Defeat Jericho

When it was time for CrossNorthPointRoadsWay Fellowship to equip their youth campus with a worship system, they knew they needed very capable equipment.

“When you have a main worship campus and a dedicated youth area, each with their own postal codes and highway offramps, you can’t wimp out,” said the church’s technical director. “Fortunately, we we get a catalog every year from that place in Indiana. It’s the same catalog that they send out at other times, only they replace the word ‘audience’ with ‘congregation,’ and ‘stage’ with ‘platform.’ That makes it appropriate for our needs.”

When asked if there was any kind of gear that was absolutely essential for the church, the technical director nodded. “Yes, we absolutely have to go with loudspeakers that come in white enclosures. That’s more important than anything. The speakers have to match the look of the space.”

CrossNorthPointRoadsWay’s Assistant Pastor For Kids 13-14 also weighed in: “To disciple our kids, we have to get them to pay attention. That’s why it’s so great to have 40,000 watts of Sack Bottom subwoofers. They really get things shaking. We can rattle a smartphone out of a kid’s hands and get them to pay attention to the REAL ‘text message,’ if you know what I mean.”

The church’s director of youth productions agreed on the importance of capable equipment. “We couldn’t possibly do work of eternal significance with less than 48 channels available at the console. We also had to have stadium-class intelligent lights. We do one very special production every year, and it’s not the same if you don’t actually have a blinding light coming down from heaven. Everything has to be top-shelf, especially when you have to outdo SouthRoadsPointCross Community Church. Not that we don’t love them as brothers and sisters, of course.”

When asked about upcoming special productions, the production director offered a few hints. “We’re going to have a series of talks on how Hollywood, the media, and pop culture in general are corrupting influences, backed up by skits and a musical featuring Iron Man, Black Widow, and Captain America.”

New Vocal Mics At SAMM

A whole slew of vocal mics debuted this year at the industry’s biggest swap-meet. Half of them would be basically indistinguishable from each other if the external styling was removed.

“We feel like the XA-58-Beta-R2D2 brings a lot of value to people,” said one rep. “Its cardioid pickup pattern isn’t all that great at rejecting feedback, but the ad copy we supply to the vendor catalogs says that it’s great for rejecting other sounds. We’re hoping that there will continue to be folks out there who don’t have a clue as to what ‘super’ and ‘hypercardioid’ patterns mean.”

New Drum Kits Announced

A new sheriff is in town, and he’s ready to clean things up around these parts.

“We originally set out to create a shellpack and snare options that would really blend well in different band situations,” said the chief designer. “We got about halfway through that process before we realized that what we really wanted to do was build a kit that could drown out everything else on stage. Drums are the foundation of the song, and the walls, and the windows, and the roof, and the paint…look, you don’t need to hear anything else. These new kits are louder than an artillery barrage, even with a Jazz player using 7As. You haven’t lived until you’ve heard ‘Nature Boy’ at 120 dB!”

We asked the celebrity endorser what he thought of the new kits. His response?

“Kill! Kill! DRUM BATTLE!”

200 Watt, All-Tube Guitar Amp Set To Debut

“It really cuts through all the wash from the bass and drumkit!” shouted the product rep.

1000 Watt, All-Tube Bass Amp Set To Debut

“It really thunders over all the wash from the guitars and drumkit!” shouted the product rep.

Get Plugged In

Ripples Audio is debuting a new series of plugins, aimed at putting powerful tools in the hands of project studios. They partnered with a renowned mix engineer to help craft each piece of software.

“It was important to us that we really capture the feel of how our endorser worked,” said a product rep. “So, the dev team went down to the studio, hung out, and took a lot of pictures. They came back, modified our main plugin suite to have more restrictive control ranges, and slapped a bunch of sexy, analog-esque graphics on the interfaces.”

We asked if users of the plugins could expect to get the same results as the endorsing engineer.

“Absolutely,” responded the representative. “If they’re in a studio with the same acoustics, and working with musicians of the same caliber, and are recording songs that sound the same, and hear things the same way that our endorser does, and have monitors that cost more than a car, then yes. Absolutely. This software package is absolutely worth the expense of $800 plus an additional $50 for a frustrating copy-protection scheme that uses unreliable hardware. It’s great. I use it at home all the time.”