Tag Archives: Console

What A Mixing Console Isn’t

Magically turning a band into something else isn’t what we’re here to do.

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

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I’m working on a new video, but it’s taking a while due to scheduling issues. (Being busy isn’t a bad thing, but still…) I figured I should put something up here to prove that I haven’t forgotten this site in the meantime.

So, in regards to a picture of a sophisticated mixing console: The device depicted is not a tool for fixing arrangement problems or interpersonal conflicts.

There, that should stir the pot a little. 🙂

Why Are Faders Labeled Like That?

Gain multipliers are hard to read.

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I’ve done a lot of typing on this site, and I’m worried that it’s getting stale – so, how about some video?

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

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

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

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

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

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

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

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

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

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

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

Let me tell ya, folks,

They all sound basically the same.

Really.

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

The Subjective Factor

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

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

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

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

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

What You’re Looking For

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

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

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

So forget about finding the unit with the best numbers.

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

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

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

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

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

The Behringer X18

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

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

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

Software Breaks The Barriers

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

Channel-per-channel dynamics.

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

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

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

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

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

Full console recall with snapshots.

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

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

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

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

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

Ultranet

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

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

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

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

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

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

The Quirk

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

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

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

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

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

Just What Signal Is It, Anyway?

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

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

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

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

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

They aren’t.

The Three Categories

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

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

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

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

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

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

The Console Problem

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

Then you move to a digital console.

Are those faders passing audio?

No.

Ah! They’re passing data that represents audio!

Nope.

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

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

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

Something’s Coming In On The Wire

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

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

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

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

It might be digital audio.

It might be control data.

It might even be both.

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

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

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

Pretty Close To An SC48

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

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

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

At FOH, they installed an SC48.

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

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

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

I’m Not Slagging The SC48

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

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

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

Full dynamics processing available on all input channels? Check.

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

A snapshot system? Check.

Recallable input gains? Check.

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

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

None of that is a mere triviality.

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

How Much Is It Worth To You?

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

I did find that functionality very useful!

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

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

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

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

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

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

Buzzkill

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

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

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

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

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

Solitude

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

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

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

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

Desperately Seeking Silence

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

Bzzzzzzzz….

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

Hmmmmmmzzzzzzzz…

Anyway.

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

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

Or, more specifically…

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

To answer that question, you need isolation.

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

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

Is the output what you expect?

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

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

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

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

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

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

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

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

How To Spend A Ton Of Money

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

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

The band’s backline was barely functional.

The guitar amps had all the midrange dialed out.

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

Right?

Why I Think Steam Machines Are Cool

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

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

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

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

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

Why is that possibility there?

The Box Is Prebuilt

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

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

The System Is Meant For Gaming

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

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

You get the idea.

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

The System Is Customizable

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

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

Practical Gain Staging For Live Sound

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

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

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

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

Distillation

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

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

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

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

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

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

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

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

The Preamp

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

They might.

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

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

A) Linear with respect to the input, and

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

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

So – what’s a decent level, then?

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

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

Channel Faders And Knobs

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

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

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

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

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

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

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

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

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

Output Masters

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

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

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

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

Post Console Processing

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

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

The Final Stage

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

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

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

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

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

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