Tag Archives: Console

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.

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

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.

So…

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.

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

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.

Anyway.

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!”

Anyway.

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.

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

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.

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

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.”


Why I Am (Not) Interested In The Industry Standard

Industry standards are helpful reference points, but are not necessarily the best possible approach.

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.

57grillsmallWant to use this image for something else? Great! Click it for the link to a high-res or resolution-independent version.

Remember my article about a patch-scheme for a “festival style” show? It actually raised an eyebrow or two. A fellow audio-human (who works on much, much, much larger shows than I do) asked me why my patch list was backwards from what everybody else does. His concern was that, in the festival situations he finds himself in, my “upside down” patch would monkeywrench things if accommodated. It would just be so much easier for everyone if I followed the industry standard of (I guess?) starting with the drums – “kick is channel 1,” in other words.

My response was that, if I had things laid out one way, and a guest engineer came in who wanted them to be another way, then I would be happy to set up any softpatch desired. What I neglected to add at the time was that, if I was “that one guy” where everyone else wanted a different order, I would be happy to just use the standard patch. It wouldn’t ruin my day at all, and it would make things easier for everybody else.

To be open and frank, though, there was something else I wanted to say. I censored myself because I think there’s a place for diplomacy and courtesy, especially when the conversation venue (Facebook comments) isn’t really good for nuance.

What I wanted to say was, “Because my way is better. Why would you put the drums first? They’re the bottom of the priority list.” (The drums are important, but in a small-venue context they usually need the least help from the PA to be in the right spot.)

What was said and unsaid in that conversation is a microcosm of how I feel about industry standards. There are industry standard mics, techniques, PA styles, stage layouts, and whatever else, and they exist for good reasons. Knowing what those reasons are is a good thing, because it’s part of understanding the craft. At the same time, though, industry standards rarely equate to “the best.” They tend to equate to “works acceptably in a wide range of situations.”

58, 57, IBM

Back when Apple Computer was struggling for acceptance, there was a saying: “Nobody every got fired for buying IBM.” IBM was the industry standard for machines used in an office environment, and even though the Macintosh computers at the time were leaps and bounds ahead in terms of user-friendliness, people kept buying IBM and compatible devices.

Why?

Because IBM was known. Large numbers of people, from the users to the admins, had experience with them. Everybody knew what to expect. They knew that appropriate software would be available, or could be developed by folks that were easy to find. They knew the parts would be there. They knew they could get work done with IBM, even if the computers weren’t revolutionary. They knew that IBM was readily respectable by everyone that they wanted to impress.

In the same way, you could say that “Nobody ever got fired for buying SM-58s and SM-57s.” They’re industry standard mics because they’re built to withstand live shows, basically sound like what they’re pointed at, and literally everybody can get them to work in a reasonable way. They’ve been around forever, and have been used by everybody, their dog, and their dog’s fleas. Even if somebody doesn’t know the model numbers, asking them to draw a picture of a vocal mic and an instrument mic will probably get you an SM-58 and an SM-57.

But they’re not the best at all times. I’ve heard a lot of 58s that imparted far too much low-mid garble to a singer’s voice, and I’ve never once easily gotten as much gain-before-feedback out of a 58 as I have an ND767a. I’ve miced up tons of amplifiers with all kinds of mics that weren’t SM-57s, and I’ve been perfectly happy about 99% of the time. I’ve done the same with drums. If “sounds decent” is the main priority, then I have a bunch of mics that do that AND take up less space than a big ol’ 57. There are other mics out there that work better for me, in terms of the total solution offered.

This isn’t to say that great things can’t happen with the SM series! I once heard an artist in a coffee shop with a keyboard amp and a 58-style mic. It was the most perfect setup for her voice that you could imagine. I wasn’t expecting what I heard, but she made it work beautifully. Sometimes, “industry standard” and “perfect for this particular application” DO line up.

My point is, though, that in a broad sense the “hidden secret” of being industry standard means being “extraordinarily average.” Thoroughly inoffensive. Safe. Something people won’t be fired for specifying and purchasing.

There’s nothing wrong with that, but for people like me…well, it’s kinda boring.

Sometimes You Need To Be Bored

That last sentence might seem a bit incendiary, depending on who you are. It’s very important to note that being un-boring is a luxury that’s unavailable to many in this business.

A good example is what happens when a venue wants to spend time working with acts that regularly tour at the regional level or above. To be acceptable to those acts (especially if they bring production techs but only minimal gear) requires that the PA and lighting rigs be easy to handle by most folks. The personnel working for the house might be excited about the new mixing consoles that lack a physical control surface, but that’s not something that everybody is prepared to accept. There are plenty of audio humans who just aren’t ready for the idea of having no physical controls at all, whereas probably every sound tech is fine with a console that has a control surface. That’s why control surfaces are still the industry standard. The new surfaceless consoles are nifty, but not for everybody, so a bit of “boring-ness” is required in order for the venue to play well with others.

Industry standards are accepted everywhere, which makes them a safe bet. Non-standards are “risky,” because they tend to conform to the desires of a smaller number of people. Risky is often exciting, however, because that’s where innovation occurs. Iterating on the standard makes the standard more refined, but it rarely produces breakthroughs. It’s entirely possible to, say, “bend the rules” on mixing console cost vs. functionality if you’re willing to do weird things (like dispense with a control surface). Some people will get it, and some people will think you’re crazy. Catering to your own brand of crazy is acceptable if, like me, a guest engineer even being in the room only happens about 0.8% of the time. It’s not acceptable at all if a band tech is going to be “driving” on a regular basis.

Why I’m Not Particularly Interested In The Industry Standard

I personally tend to shrug my shoulders at industry standards for the same reason that people shrug their shoulders in general: There’s almost nothing exciting about what’s been done a million times. Since I currently don’t have to meet riders or provide an easy environment for other techs to work in, I have the luxury of basically doing whatever I want as long as it works.

I love giving “upstarts” and bargain items a chance, because it’s fun to see just how far a piece of gear can go if you spend some time with it.

I don’t fight feedback with per-mix graphic EQs, because the idea of hacking up a whole mix to solve a problem with one input seems crazy to me.

I use a homebrew console because I wanted to have a virtual, independent monitor-world, and nobody made a traditional console I could afford that would do that in the way I wanted.

I don’t use a control surface for mixing because I’ve never cared about moving a whole bunch of faders at once.

I’ve never personally owned an SM-58 or 57, because they just aren’t interesting to me.

I’ve stuffed a cheap measurement mic inside a kick drum on several occasions, because I wanted to see how it would work. (It was actually pretty okay.)

And I just generally roll my eyes at how so much of show production, which used to be a kind of “outlaw” business that pushed boundaries and did things for the fun of it, has become a beige, corporatized affair of trying to basically be like everybody else. It’s like cars, you know? They used to be cool, distinctive works of art, and now every car company is essentially making the same three boring-as-dirt sedans, three bland SUVs, and three unremarkable pickup trucks, because it’s all run by “money” people now who are terrified of not being more profitable next quarter and thus will never do anything interesting YOU GUYS LET ME KNOW IF I’M RAMBLING, ‘KAY?

Now, you can bet that, if I ever went to work at an AV company or production provider, I would be willing to conform to industry standards. In that environment, that would be the appropriate thing to do.

But right now, I have the freedom to be weird and have fun – so I intend to enjoy myself.

I’ll say it again. “Industry standard” doesn’t necessarily mean “the best.” It just means “people will accept this about 95% of the time.”


Not Remotely Successful

Just getting remote access to a mix rig is not a guarantee of being able to do anything useful with that remote access.

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.

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

The nature of experimentation is that your trial may not get you the expected results. Just ask the rocket scientists of the mid-twentieth century. Quite a few of their flying machines didn’t fly. Some of them had parts that flew – but only because some other part exploded.

This last week, I attempted to implement a remote-control system for the mixing console at my regular gig. I didn’t get the results I wanted, but I learned a fair bit. In a sense, I think I can say that what I learned is more valuable than actually achieving success. It’s not that I wouldn’t have preferred to succeed, but the reality is that things were working just fine without any remote control being available. It would have been a nice bit of “gravy,” but it’s not like an ability to stride up to the stage and tune monitors from the deck is “mission critical.”

The Background

If you’re new to this site, you may not know about the mix rig that I use regularly. It’s a custom-built console that runs on general computing hardware. It started as a SAC build, but I switched to Reaper and have stayed there ever since.

To the extent that you’re talking about raw connectivity, a computer-hosted mix system is pre-primed for remote control. Any modern computer and accessible operating system will include facilities for “talking” to other devices over a network. Those connectivity facilities will be, at a basic level, easy to configure.

(It’s kind of an important thing these days, what with the Internet and all.)

So, when a local retailer was blowing out 10″ Android tablets for half price, I thought, “Why not?” I had already done some research and discovered that VNC apps could be had on Android devices, and I’ve set up VNC servers on computers before. (It’s not hard, especially now that the installers handle the network security configuration for you.) In my mind, I wasn’t trying to do anything exotic.

And I was right. Once I had a wireless network in place and all the necessary software installed, getting a remote connection to my console machine was as smooth as butter. Right there, on my tablet, was a view of my mixing console. I could navigate around the screen and click on things. It all looked very promising.

There’s a big difference between basic interaction and really being able to work, though. When it all came down to it, I couldn’t easily do the substantive tasks that would make having a remote a handy thing. It didn’t take me long to realize that tuning monitors while standing on the deck was not something I’d be able to do in a professional way.

A Gooey GUI Problem

At the practical level, the problem I was having was an interface mismatch. That is, while my tablet could display the console interface, the tablet’s input methodology wasn’t compatible with the interface being displayed.

Now, what the heck does that mean?

Reaper (and lots of other audio-workstation interfaces) are built for high-precision pointing devices. You might not think of a mouse or trackball as “high precision,” but when you couple one of those input devices with the onscreen pointer, high precision is what you get. The business-end of the pointer is clearly visible, only a few pixels wide, and the “interactivity radius” of the pointer is only slightly larger. There is an immediately obvious and fine-grained discrimination between what the pointer is set to interact with, and what it isn’t. With this being the case, the software interface can use lots of small controls that are tightly packed.

Additionally, high-precision pointing allows for fast navigation across lots of screen area. If you have the pointer in one area of the screen and invoke, say, an EQ window that pops open in another area, it’s not hard to get over to that EQ window. You flick the mouse, your eye finds the pointer, you correct on the fly, and you very quickly have control localized to the new window. (There’s also the whole bonus of being able to see the entire screen at once.) With high-precision input being available, the workstation software can make heavy use of many independent windows.

Lastly, mice and other high-precision pointers have buttons that are decoupled from the “pointing” action. Barring some sort of failure, these buttons are very unambiguous. When the button is pressed, it’s very definitely pressed. Clicks and button holds are sharply delineated and easily parsed by both the machine and the user. The computer gets an electrical signal, and the user gets tactile feedback in their fingers that correlates with an audible “click” from the button. This unambiguous button input means that the software can leverage all kinds of fine-grained interactions between the pointer position and the button states. One of the most important of those interactions is the dragging of controls like faders and knobs.

So far so good?

The problem starts when an interface expecting high-precision pointing is displayed on a device that only supports low-precision pointing. Devices like phones and tablets that are operated by touch are low-precision.

Have you noticed that user interfaces for touch-oriented devices are filled with big buttons, “modal” elements that take over the screen, and expectations for “big” gestures? It’s because touch control is coarse. Compared to the razor-sharp focus of a mouse-driven pointer, a finger is incredibly clumsy. Your hand and finger block a huge portion of the screen, and your finger pad contacts a MASSIVE area of the control surface. Sure, the tablet might translate that contact into a single-pixel position, but that’s not immediately apparent (or practically useful) to the operator. The software can’t present you with a bunch of small subwindows, as the miniscule interface elements can’t be managed easily by the user. In addition, the only way for the touch-enabled device to know the cursor’s location is for you to touch the screen…but touch, by necessity, has to double as a “click.” Interactions that deal with both clicks and movement have to be forgiving and loosely parsed as a result.

Tablets don’t show big, widely spaced controls in a single window because it looks cool. They do it because it’s practical. When a tablet displays a remote interface that’s made for a high-precision input methodology, life gets rather difficult:

“Oh, you want to display a 1600 x 900, 21″ screen interface on a 1024 X 600, 10″ screen? That’s cool, I’ll just scale it down for you. What do you mean you can’t interact with it meaningfully now?”

“Oh, you want to open the EQ plugin window on channel two? Here you go. You can’t see it? Just swipe over to it. What do you mean you don’t know where it is?”

“Oh, you want to increase the send level to mix three from channel four? Nice! Just click and drag on that little knob. That’s not what you touched. That’s also not what you touched. Try zooming in. I’m zoomi- wait, you just clicked the mute on channel five. Okay, the knob’s big now. Click and drag. Wait…was that a single click, or a click and hold? I think that was…no. Okay, now you’re dragging. Now you’ve stopped. What do you mean, you didn’t intend to stop? You lifted your finger up a little. Try again.”

With an interface mismatch, everything IS doable…but it’s also VERY slow, and excruciatingly difficult compared to just walking back to the main console and handling it with the mouse. Muting or unmuting a channel is easy enough, but mixing monitors (and fighting feedback) requires swift, smooth control over lots of precision elements. If the interface doesn’t allow for that, you’re out of luck.

Control States VS. Pictures Of Controls

So, can systems be successfully operated by remotes that don’t use the same input methodology as the native interface?

Of course! That’s why traditional-surface digital consoles can be run from tablets now. The tablet interfaces are purpose-built, and involve “state” information about the main console’s controls. My remote-control solution didn’t include any of that. The barrier for me is that I was trying to use a general-purpose solution: VNC.

With VNC, the data transmitted over the network is not the state of the console’s controls. The data is a picture of the console’s controls only, with no control-state data involved.

That might seem confusing. You might be saying, “But there is data about the state of the controls! You can see where the faders are, and whether the mutes are pressed, and so on.”

Here’s the thing, though. You’re able to determine the state of the controls because you can interpret the picture. That determination you’ve made, however, is a reconstruction. You, as a human, might be seeing a picture of a fader at a certain level. Because that picture has a meaning that you can extract via pattern recognition, you can conceptualize that the fader is in a certain state – the state of being at some arbitrary level of gain. To the computer, though, that picture has no meaning in terms of where that fader is.

When my tablet connects to the console via VNC, and I make the motions to change a control’s state, my tablet is NOT sending information to the console about the control I’m changing. The tablet is merely saying “click at this screen position.” For example, if clicking at that screen position causes a channel’s mute to toggle, that’s great – but the only machine aware of that mute, or whether that mute is engaged or disengaged, is the console itself. The tablet itself is unaware. It’s up to me to look at the updated picture and decide what it all means…and that’s assuming that I even get an updated picture.

The cure to all of this is to build a touch-friendly interface which is aware of the state of the controls being operated. You can present the knobs, faders, and switches in whatever way you want, because the remote-control information only concerns where that control should be set. The knobs and faders sit in the right place, because the local device knows where they are supposed to be in relation to their control state. Besides solving the “interface mismatch” problem, this can also be LIGHT YEARS more efficient.

(Disclaimer: I am not intimately aware of the inner workings of VNC or any console-remote protocol. What follows are only conjectures, but they seem to be reasonable to me.)

Sending a stream of HD (or near HD) screenshots across a network means quite a lot of data. If you’re using jpeg-esque compression, you can crush each image down to 100 kilobytes and still have things be usable. VNC can be pretty choosy about what it updates, so let’s say you only need one full image every second. You won’t see meters move smoothly or anything like that, but that’s the price for keeping things manageable. The data rate is about 819 kbits/ second, plus the networking overhead (packet headers and other communication).

Now then. Let’s say we’ve got some remote-control software that handles all “look and feel” on the local device (say, a tablet). If you represent a channel as an 8-bit identifier, that means you can have up to 256 channels represented. You don’t need to actually update each channel all the time to simply get control. Data can just be sent as needed, of course. However, if you want to update the channel meters 30 times per second, that meter data (which could be another 8-bit value) has to be attached to each channel ID. So, 30 times a second, 256 8-bit identifiers get 8-bits of meter information data attached to each of them. Sixteen bits multiplied by 256 channels, multiplied by 30 updates/ second works out to about 123 kbits/ second.

Someone should check my math and logic, but if I’m right, nicely fluid metering across a boatload of channels is possible at less than 1/6th the data rate of “send me a screenshot” remote control. You just have to let the remote device handle the graphics locally.

Control-state changes are even easier. A channel with fader, mute, solo, pan, polarity, a five-selection routing matrix, and 10 send controls needs to have 20 “control IDs” available. A measly little 5-bit number can handle that (and more). If the fader can handle 157 “integer” levels (+12 dB to -143 dB and “-infinity”) with 10 fractional levels of .1 dB between each integer (1570 values total), then the fader position can be more than adequately represented by an 11-bit number. If you touch a fader and the software sends a control update every 100th of a second, then a channel ID, control ID, and fader position have to be sent 100 times per second. That’s 24 bits multiplied by 100, or 2.4 kbits/ second.

That’s trivial compared to sending screenshots across the network, and still almost trivial when compared to the “not actually fast” data rate required to update the meters all the time.

Again, let me be clear. I don’t actually know if this is how “control state” remote operation works. I don’t know how focused the programmers are on network data efficiency, or even if this would be a practical implementation. It seems plausible to me, though.

I’m rambling at this point, so let me tie all this up: Remote control is nifty, and you can get the basic appearance of remote control with a general purpose solution like VNC. If you really need to get work done in a critical environment, though, you need a purpose built solution that “plays nice” at both the local and remote ends.


A Vocal Group Can Be Very Helpful

Microsurgery is great, but sometimes you need a sledgehammer.

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.

Folks tend to get set in their ways, and I’m no exception. For ages, I have resisted doing a lot of “grouping” or “busing” in a live context, leaving such things for the times when I’ve been putting together a studio mix. I think this stems from wanting maximum flexibility, disliking the idea of hacking at an EQ that affects lots of inputs, and just generally being in a small-venue context.

Stems. Ha! Funny, because that’s a term that’s used for submixes that feed a larger mix. Submixes that are derived from grouping/ busing tracks together. SEE WHAT I DID THERE?

I’m in an odd mood today.

Anyway…

See, in a small-venue context, you don’t often get to mix in the same way as you would for a recording. It’s often not much help to, say, bus the guitars and bass together into a “tonal backline” group. It’s not usually useful because getting a proper mix solution so commonly comes down to pushing individual channels – or just bits of those channels – into cohesion with the acoustic contribution that’s already in the room with you. That is, I rarely need to create a bed for the vocals to sit in that I can carefully and subtly re-blend on a moment’s notice. No…what I usually need to do is work on the filling in of individual pieces of a mix in an individual way. One guitar might have its fader down just far enough that the contribution from the PA is inaudible (but not so far down that I can’t quickly push a solo over the top), while the other guitar is very much a part of the FOH mix at all times.

The bass might be another issue entirely.

Anyway, I don’t need to bus things together for that. There’s no point. What I need to do for each channel is so individualized that a subgroup is redundant. Just push ’em all through the main mix, one at a time, and there you go. I don’t have to babysit the overall guitar/ bass backline level – I probably have plenty already, and my main problem is getting the vocals over the whole thing anyway.

The same overall reasoning works if you’ve only got one vocal mic. There’s no reason to chew up a submix bus with one vocal channel – I mean, there’s nothing there to “group.” It’s one channel. However, there are some very good reasons to bus multiple vocal inputs into one signal line, especially if you’re working in a small venue. It’s a little embarrassing that it’s taken me so long to embrace this thinking, but hey…here we are NOW, so let’s go!

The Efficient Killing Of Feedback Monsters

I’m convinced that a big part of the small venue life is the running of vocal mics at relatively high “loop gain.” That is, by virtue of being physically nearby to the FOH PA (not to mention being in an enclosed and often reflective space) your vocal mics “hear” a lot more of themselves than they might otherwise. As such, you very quickly can find yourself in a situation where the vocal sound is getting “ringy,” “weird,” “squirrely,” or even into full-on sustained feedback.

A great way to fight back is a vocal group with a flexible EQ across the group signal.

As I said, I’ve resisted this for years. Part of the resistance came from not having a console that could readily insert an EQ across a group. (I can’t figure out why the manufacturer didn’t allow for it. It seems like an incredibly bizarre limitation to put on a digital mixer.) Another bit of my resistance came from not wanting to do the whole “hack up the house graph” routine. I’ve prided myself on having a workflow where the channel with the problem gets a surgical fix, and everything else is left untouched. I think it’s actually a pretty good mentality overall, but there’s a point where a guy finally recognizes that he’s sacrificing results on the altar of ideology.

Anwyay, the point is that a vocals-only subgroup with an EQ is a pretty good (if not really good) compromise. When you’ve got a bunch of open vocal mics on deck, the ringing in the resonant acoustical circuit that I like to call “real music in a real room” is often a composite problem. If all the mics are relatively close in overall gain, then hunting around for the one vocal channel that’s the biggest problem is just busywork. All of them together are the problem, so you may as well work on a fix that’s all of them together. Ultra-granular control over individual sources is a great thing, and I applaud it, but pulling 4 kHz (or whatever) down a couple of dB on five individual channels is a waste of time.

You might as well just put all those potential problem-children into one signal pipe, pull your offending frequency out of the whole shebang, and be done with the problem in a snap. (Yup, I’m preaching to myself with this one.)

The Efficient Addition Of FX Seasoning

Now, you don’t always want every single vocal channel to have the same amount of reverb, or delay, or whatever else you might end up using. I definitely get that.

But sometimes you do.

So, instead of setting multiple aux sends to the same level, why not just bus all the vocals together, set a pleasing wet/ dry mix level on the FX processor, and be done? Yes, there are a number of situations where you should NOT do this: If you need FX in FOH and monitor world, then you definitely need a separate, 100% “wet” FX channel. (Even better is having separate FX for monitor world, but that’s a whole other topic.) Also, if you can’t easily bypass the FX chain between songs, you’ll want to go the traditional route of “aux to FX to mutable return channel.”

Even so, if the fast and easy way will work appropriately, you might as well go the fast and easy way.

Compress To Impress

Yet another reason to bus a bunch of vocals together is to deal with the whole issue of “when one guy sings, it’s in the right place, but when they all do a chorus it’s overwhelming.” You can handle the issue manually, of course, but you can also use compression on the vocal group to free your attention for other things. Just set the compressor to hold the big, loud choruses down to a comfortable level, and you’ll be most of the way (if not all the way) there.

In my own case, I have a super-variable brickwall limiter on my full-range output, a limiter that I use as an overall “keep the PA at a sane level” control. A strategy that’s worked very well for me over the last while is to set that limiter’s threshold as low as I can possibly get away with…and then HAMMER the limiter with my vocal channels. The overall level of the PA stays in the smallest box possible, while vocal intelligibility remains pretty decent.

Even if you don’t have the processing flexibility that my mix rig does, you can still achieve essentially the same thing by using compression on your vocal group. Just be aware that setting the threshold too low can cause you to push into feedback territory as you “fight” the compressor. You have to find the happy medium between letting too little and too much level through.

Busing your vocals into a subgroup can be a very handy thing for live-audio humans to do. It’s surprising that it’s taken me so long to truly embrace it as a technique, but hey – we’re all learning as we go, right?


Holistic Headroom

If you have zero headroom anywhere, you have zero headroom everywhere.

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.

“Headroom” is a beloved buzzword for audio craftspersons. Part of the reason it’s beloved is because you can blame your problems on the lack of it:

“I hate those mic pres. They don’t have enough headroom.”

“I’m always running out of headroom on that console’s mix buses.”

“I need to buy a more powerful amplifier for my subs, because I this one doesn’t have enough headroom.”

(I’m kinda tipping my hand a bit with that last one, in terms of this post being sort of a “follow on” to my article about clipping.)

Headroom is sometimes treated as a nebulous sort of concept – a hazy property that really good gear has enough of, and not-so-good gear doesn’t possess in the required quantity. In my opinion, though, headroom is pretty easy to define, and its seeming mysteriousness is due to it being used as a “blamecatcher” for things that didn’t go as planned.

Headroom, as I was taught, is “the difference between the maximum attainable level and the nominal level.” In other words, if a device can pass a signal of greater intensity than is required for a certain situation, then the device has some non-zero amount of headroom. For example, if your application requires a console’s main bus to pass 0 dBu (decibels referenced to 0.775 volts, RMS), and the console can pass +24 dBu, then you have 24 dB of headroom in the console.

(If it’s available, and ya ain’t usin’ it, it’s headroom.)

The overall concept is pretty easy to understand, but what a good number of folks aren’t taught, and often fail to realize for a good long while (this includes me), is that headroom is holistic, and “lowest common denominator.” That is to say:

Two or more audio components – whether electrical or acoustical – connected together all have the SAME effective headroom, and that effective headroom is equal to the LOWEST amount of headroom available at any point in the signal chain.

So…what the heck does that mean?

Everything Has A Maximum Level – Everything

To start with, it’s important to point out that hyphenated bit in the above definition. Especially because this is a site about live-performance, what you have to realize is that absolutely everything connected to that live performance has a maximum amount of appropriate signal intensity. Even acoustical sources and your audience qualify for this. Think about it:

A singer can’t sing any louder than they can sing.

A mic can only handle so much SPL.

A preamp can only swing a limited amount of voltage at its outputs.

Different parts of a console’s internal signal path have limits on how much signal they can handle.

A power amplifier can’t deliver an infinite amount of voltage.

Speakers handle a limited amount of power.

The people listening to the show have a finite tolerance for sound pressure.

…and every single one of these “components” is connected to the others. Sure, the connection may not be a direct, electrical hookup, but the influences of other parts of the system are still felt. If your system can create a “full tilt boogie” sound pressure level of 125 dB SPL C, but your audience will only tolerate about 105, then that lower level becomes your “don’t exceed” point. Go beyond it, and you effectively “clip” the audience…which makes your 20 dB of unused PA capability partially irrelevant. That leads to my next point.

Your Minimum Actual Headroom Is All You Effectively Have

Sometimes, a singer will “run out of gas.” They may have strained themselves, or they might not be feeling well, or they might just be tired. As a result, their maximum acoustical output drops by some amount.

Here’s the thing.

The entire system’s EFFECTIVE headroom has just dropped by that amount. If the singer is 10 dB quieter than they used to be, you’ve just lost 10 dB of effective headroom.

Now – before you start getting bent out of shape, complaining that your console’s mix bus headroom hasn’t magically changed, look at that paragraph again. The key is the word “effective.”

Of course your console can still pass its maximum signal. Of course your loudspeakers still handle the same power as they did a moment ago. As isolated components, their absolute headroom has not changed in any way.

But components working in a complete electro-acoustical system are not isolated, and are therefore limited by each other in various ways.

In the case of a singer getting worn out, their vocal “signal” drops closer to the noisefloor of the band playing around them. Now, if we were talking about an electrical device, the noisefloor staying the same with a decrease in maximum level above that noisefloor would be – what? Yes: A loss of headroom.

The way this affects everything else is that you now have to drive the vocal harder to get a similar mix. (It’s not the same mix, because there’s less acoustical separation between the singer and the band at the point of the mic capsule, but that’s a different discussion.) Because the singer’s overall level has dropped, your gain change might not be pushing you any closer to clipping an electrical device…but you are definitely closer to the point where your system will “ring” with feedback. A system in feedback, effectively, has reached its maximum available output.

Your effective headroom has dropped.

A Bigger Power Amp Isn’t Enough

Okay – here’s the bit that’s directly related to my “clipping” article.

The concept of holistic headroom is one of the larger and fiercer bugaboos to be found in the piecing together of live-audio rigs. As many bugaboos do, it grows to a fearsome size by feeding on misconceptions and mythology. There is a particular sub-species of this creature that’s both common and venomous: The idea that a system headroom problem can be fixed by purchasing more powerful amplifiers.

Now, if you’re constantly clipping your amps because the system won’t get loud enough for your application, then yes, you need to do something about the problem. However, what you need to do has to be effective on the whole, and not just for one isolated part of the signal chain. Buying a bigger amplifier will probably get you some headroom at the amplifier, but it might not actually get you any more effective headroom (which is what actually matters). If your old amplifier’s maximum level was equal to your speakers’ power handling, and the new amplifier is more powerful than the old one, then you’ve done nothing in terms of effective headroom.

The loudspeakers were already hitting their maximum level. As such, they had zero headroom, and your new amp is thus effectively limited to zero additional headroom. Your enormously powerful amp is doing virtually nothing for you, except for letting you hit your unchanged maximum level without seeing clip lights.

To be fair, the system will get somewhat louder because loudspeakers don’t “brickwall” at their maximum input levels. Also, the nature of most music is that the peaks are significantly higher than the continuous level, which lets you get away with a too-big amp for a while. You will get some more level for a while, but your speakers will die much sooner than they should – and when they do, your system will become rather quieter…

Anyway.

The point is that, if you want a system headroom increase of “x” decibels, then you have to be sure that every part of your system – not just one piece – has “x” more decibels to give you. If you’re going to get more power, you have to make sure that you also have that much more “speaker” to receive that power. (And this gets into all kinds of funny business, like whether or not you can buy speakers that are just as efficient as what you’ve had while handling more power, or whether you need to buy more of the same speakers, and if that’s a good idea because of arrayability, or…)

There’s also the question of whether or not a more powerful system is what your audience even wants. It all ties together, because headroom is holistic.


Mixing For The Stream

The sound for a stream and the sound for an in-room audience have competing priorities.

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.

Over the last several weeks, I’ve had the pretty-darn-neat job of mixing for livestreamed shows. AMR.fm is doing these live broadcasts on Monday nights, broadcasts that include Q&A with bands as well as live music.

It’s pretty nifty, both as an event and as a technical exercise. Putting your best foot forward on a live stream isn’t a trivial thing, but a big part of having fun is rising to a challenge, right?

Right?

Oh, come on. Don’t look at me like that. You know that challenges are where the serious enjoyment is. (Unless the challenge is insurmountable. Then it’s not so fun.)

Anyway.

The real bugaboo of doing an actual, honest-to-Pete live gig that’s also being streamed is that you have at least two different audiences, each with different priorities. To keep them all happy, you need to be able to address each separate need independently (or quasi-independently, at least). I use the word “need” because of one particular reality:

In a small-venue, the needs of the show in the room are often contrary to the needs of the show on the stream.

One way that is manifests in practical terms is that…

You Probably Don’t Want A Board Feed

“Board Feeds” can be wondrous things. In a large venue, with reasonable stage-volume, there’s a real chance that everything is in the PA, and at “full range.” That is to say, the mix includes all the instruments (even the loud ones), and the tonal shaping applied to each input is only minimally influenced by the acoustic contribution from the stage. The PA is being used to get the ENTIRE band out to the audience, and not just to fill in the spaces where a particular input isn’t at the right volume.

In the above scenario, taking a split from the main mix (before loudspeaker processing) could be a great and easy option for getting audio to stream out.

In a small venue, though, things can be rather more tricky.

I’ve written about this before. In a small room, putting everything in the PA is often unnecessary…and also a bad idea. It’s very possible to chase everybody out with that kind of volume. Rather, it’s desirable to only use the PA for what’s absolutely necessary, and ignore everything else. The “natural” acoustical contribution from the band, plus a selective contribution from the PA come together into a total acoustic solution that works for the folks in the room.

The key word there is “acoustic.”

A small-venue board feed to a live stream is often the wrong idea, because that feed is likely to sound VERY different than what’s actually in the room. The vocals might be aggressively high-passed. The guitar amps might not be present at all. The drums might sound very odd, and be very low in the mix.

And it’s all because the content of that feed is meant to combine with acoustic events to form a pleasant whole. Unfortunately, in this situation, a board-feed plus nothing is lacking those acoustical events, and so the stream sounds terrible.

The Right Mix For The Right Context

Obviously, you don’t want the stream to sound bad, or even just “off.” So – what can you do? There are two major options:

1) Capture the total acoustical event in the room, and stream that.

2) Have a way to create an independent mix for the stream that includes everything, and in a natural tonality.

The first option is easy, and often inexpensive, but it rarely sounds all that great. Micing a room, even in stereo, can be pretty “hit or miss.” Sure, a nice stereo pair in a symphony hall is likely to sound pretty good, but most folks aren’t playing symphonies in a concert hall to a quiet crowd. As likely as not, you’re streaming some kind of popular music style that’s taking place in a club, and the crowd is NOT being quiet.

Now, even with all that, there’s nothing wrong with taking the first option if it’s all you’ve got. I’ve personally enjoyed my fair share of concert videos that are nothing more complex than “micing the room.” Still, why not reach higher if you can?

Trying for something better requires some kind of “broadcast split.” There are different ways to make it happen, but the most generally feasible way is likely the route that I’ve chosen: Connect each input to two separate mix rigs. A simple splitter snake and a separate “stream mix” console are pretty much what you need to get started.

The great thing about using a separate console for the broadcast is that you have the freedom to engage in all kinds of weirdness on either console (live or stream), without directly affecting the other mix. Need a “thin” vocal in the room, but a rich and full tone for the stream? No problem! Do the guitar amps need no help from the PA, but do need to be strongly present for the broadcast audience? No sweat! Having separate consoles means that the “in-studio” audience and the stream listeners can both be catered to, without having to completely sacrifice one group on the other’s altar.

Having a totally separate mix for the broadcast is not without its own challenges, though. It would be irresponsible for me to forget to point out that mixing for two, totally separate audiences can be a real workout. If you’re new to audio, you might want to have a different person handle one mix or the other. (I’m not new to being a sound human, but I still have to cope by giving neither the live nor the broadcast mixes my full attention. I take every shortcut I can on “broadcast day,” and I let plenty of things just roll along without correction for much longer than I usually would.) Even with separate mix rigs, the broadcast mix is still partially (though indirectly) affected by the acoustical events in the room – like “ringy” monitors on deck. That being so, any “live” problem you have is likely to be VERY audible to the broadcast audience. If you’re the only one around to manage it all, that’s fine…but be ready.

I should also mention that having some way to do “broadcast levelling” on the stream feed is a good idea. Especially in my case, where we transition from Q&A to music, the dynamic range difference involved can be pretty startling. To the folks in the room, the dynamic swing is expected to some degree. To the stream listeners, though, having to lunge for the volume control isn’t too pleasant. One way to create a broadcast leveller is to insert a brickwall (infinity:1, zero attack) limiter with a long (say, five seconds) release time across the entire broadcast mix. You then set the threshold and output gain so as to minimize the difference between the loud and soft portions of the program. Using automatic levelling does sound a bit odd versus doing it manually, but it can free up your attention for other things at times.

Then again, automatic levelling does require you to do more to manage your broadcast-mix channel mutes, because a side effect of making everything “the same amount of loud at all times” means that your noise floor gets CRANKED.

…but hey, if this gig wasn’t interesting, we wouldn’t want it, right?


The Order Matters

Getting your signal chain sorted out is key – especially when monitor world and FOH come together.

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.

Sometimes, you have to do things that “break the rules.”

Audio-humans internalize a lot of pointers as they learn their craft, and those tactics are often in place for very good reasons. When a given way of making things happen has survived for decades, it’s usually because it’s either a really good idea, or we just haven’t found a way around it yet. The problem that arises, though, is that a lot of techs don’t know the “deep roots” of why certain signal flows are as they are.

For instance, just about everybody knows that a gate should – 99% of the time – be placed pre-compression. Not everybody can verbalize the “why” of that rule, though. The “deep root” of the rule is that dynamic range expansion (gating) works more effectively as the dynamic range of an input signal increases. The less of a level difference that you have between the material you want gated out and the material you want to keep, the less able you are to cause the gate to discriminate between the two. Compressing a signal at some point that’s pre-gate is just working against yourself, because compression is dynamic range reduction.

But I digress.

The point of this article isn’t to get into every kind of signal flow arrangement. The idea here is to relate an anecdote that shows why I had to “break some rules” recently. It was all in the name of getting FOH (Front Of House) and monitor world to play nicely.

FX Out Front and On Deck

As I was soundchecking a band, one of the players expressed a request to have reverb on his instrument. He also specifically requested that the reverb be routed to the monitors.

Here’s where the trouble can start.

See, “everybody knows” that reverbs are fed from post-fader sends. Most of the time, this is the right thing to do. You use the send to create a reverb proportionality, and if you end up pushing the channel level around, the proportionality stays the same. If the fader goes up 6 dB, the reverb level goes up 6 dB – the wet/ dry mix remains as it was set. That’s a good thing.

Except when it isn’t.

The problem in the “Curious Case of a Reverb That’s Going to FOH and Monitor World” is that you DON’T want the reverb level to track with level changes out front. If it does, then the wet/ dry blend on deck can go all over the place during the show. This is especially true in small venues, where a instrument may be completely “out” until a solo, at which point you drive the level up into audible territory. That could mean an effective dynamic range of 80 db or more. Possibly a lot more.

Obviously, appearing and disappearing reverb isn’t what the gents on stage are after. As a result, the “post-fader sends to FX” rule has to go out the window, because it’s no longer appropriate. Instead, the reverb has to be run from a pre-fader send. As long as you don’t fiddle with your preamp gain, the reverb level will be unaffected by what you’re doing out front.

Or will it?

The other thing you have to be aware of is where that pre-fader send lives in relation to your channel EQ. If you have something bizarre going on with the channel EQ for FOH (and you very well might), and that pre-fader send takes a split AFTER the EQ, your reverb may sound awfully strange.

What To Do, What To Do?

The first thing that you have to do is prioritize. In most cases, making a consistent blend “easy” for monitor world should come before making FOH easy. (There’s probably a whole article to be written about this, but the short version is that you can often hear, and act on, issues in FOH faster than issues on deck.)

The next thing to do is to figure out what you need for that prioritization to be fulfilled. In this case, I needed reverb that was driven from a pre-fader, pre-EQ signal. I also needed the “wet” audio from the reverb to be independently routable to FOH and the monitor wedges. Making this happen for me is no problem, because I run a console with insanely flexible routing. I can actually use “subchannels” within channels to pass audio “around” processors, and any channel can send to or receive from any other channel. I also have the built-in option to run sends pre or post any channel processing.

But, what if you don’t have all that?

Heck, what if you don’t have completely separate sets of channels for FOH and monitor land?

You can still make this happen. Take a look:

The “half-jacked” insert lets you mult (split) the original signal over to the reverb. At the same time, the signal continues to flow through the FOH channel and its monitor sends. You can then take the reverb processor’s output, put that in a different channel, and use the pre-fader sends to get reverb to monitor world. The reverb channel’s fader output can then be blended into FOH as necessary.

With this kind of setup, you can go hog-wild with your FOH levels, and monitor world won’t be directly affected. There are other ways of accomplishing this, of course, but this setup is one of the simpler ones.

Yes, this is a bit more complicated than what you might think of “off the cuff,” but it lets you have what you need out front without compromising what the folks on deck can have for themselves. I think it’s worth doing if you have the channels, and it’s not that hard to adapt to your own needs…

…you just have to remember that “the order matters.”