If terms such as gain structure, impedance matching and headroom are unfamiliar, or worse, give you a headache, don’t worry, you’re not alone.
Most church sound techs would rather have their gear work perfectly right out of the box than have to tweak it into compliance.
Nevertheless, when it comes to setting up and operating a sound system, a working knowledge of gain structure (and a few related concepts) will help you get the best possible performance from whatever equipment you use.
In short, gain structuring has to do with setting the relative levels of audio signals going into and out of two or more connected audio circuits.
Audio gear has a range of input and output signal levels within which it sounds good. Going outside of that range results in problems such as hiss, distortion, reduced fidelity (especially when dealing with digital gear) and lowered power output.
The goal here is to provide you with a basic understanding of gain structure in a live sound system and will show you some ways to optimize gain structure for each piece of gear in the signal chain.
As an aid to understanding, see the diagram of a signal chain in Figure 1. Some sound systems will be simpler than the one depicted in the diagram, while others will be more complex, but the basic principles apply to any configuration.
Maximum Headroom
All audio gear has a peak maximum signal level (above which the signal begins clipping) and what is referred to as its noise floor: the natural noise of the electronics when no input signal is present (see Figure 2).
The total difference between the two extremes is called the dynamic range, which is expressed in decibels. For example, if the peak maximum signal level of a device is +24 dBu and the noise floor is -60 dBu, the device has a dynamic range of 84 dB (24 dB + 60 dB = 84 dB).
The difference between the noise floor and the nominal level at which gear operates (+4 dBu on a typical VU meter) is called the signal-to-noise ratio (S/N ratio).
Finally, the difference between the nominal operating level and the maximum peak level is referred to as headroom.
Why is all that important, and what does it have to do with getting optimal performance from your audio gear? As a rule, you want to drive the inputs of a piece of gear at as high a level as possible without inducing distortion.
So if the nominal level is +4 dBu and the peak maximum level is +24 dBu, theoretically you have 20 dB of headroom to work with.
That means you’ll probably want to set the input level so that the absolute loudest peaks fall a few decibels short of the maximum, say around +16 dBu on a PPM scale (Peak Program Meter).
I say probably because practice doesn’t always follow theory — use your ears as the final judge. Conversely, input signals that fall far below the nominal level become increasingly noisy as they approach the noise floor.
Setting the optimal output level is even trickier, because output stage characteristics can vary wildly from one piece of gear to another.
For example, some begin distorting at relatively low settings, while others sound best when they’re wide open.
That’s one of the reasons it’s important to become familiar with each piece of gear in the signal chain. If one or more pieces of digital gear are in the chain, additional considerations will apply.
That may seem plain enough, and in general it is, but a number of additional variables must be considered at each link in the chain.
And only by recognizing and dealing with those variables can you get optimal performance from your sound system. Let’s take a closer look…
The Source
When structuring gain relationships, you should always start at the beginning of the signal chain, which is, not surprisingly, the sound source.
In the case of direct line feeds from sources like instrument amplifiers, electronic keyboards and personal mixers on stage, a sound tech can do little more than request that the signals arrive at the mixer inputs at optimal levels.
But the tech does have control over at least two sources: condenser microphones with built-in pads and direct injection (DI) boxes with selectable output levels.
Many condenser and RF microphones have a built-in pad (input attenuator) that reduces the signal between the capsule and the output electronics by 10 dB or so. Generally, you won’t need to engage the pad unless the mic is used on an especially loud sound source.
For example, if an AKG C 535 condenser mic is used on a loud vocalist or snare drum, then its 14-dB pad should be engaged. If a similar condenser mic is used on acoustic strings or as an overhead on a drum kit, the pad can stay off.
Engaging the pad for soft sound sources raises the noise floor of the capsule to a point where it can be noticeable during quiet passages; so use the pad only when necessary.
The general rule of thumb is: if you hear something that sounds like clipping or limiting from a condenser or RF mic itself, activate the mic pad. If not, then full steam ahead.
Many DI boxes have selectable output levels. For instance, DOD 265 Stagehand DI boxes have selectable 20-dB and 40-dB output pads, and some Whirlwind DI boxes have 20-dB pads.
Because many mixing consoles have pads on the input strips, it’s best to send as hot a signal as possible from a DI box without clipping the output of the box itself (something that usually happens only on active DI boxes).
That lets you trim back the signal to something usable at the console input while keeping the signal as hot as possible for its trip through the signal snake to the console.
This procedure helps attenuate the effects of any ground-loop problems that might exist on that line due to its interaction with, say, a bass guitar amplifier on stage.
A quick note regarding ground loops: If a passive DI box has a metal XLR jack and the mic cable’s shield isn’t properly disconnected from the XLR shell, then it’s impossible to break a ground loop using the ground-lift switch on the DI box.
In that case, you’ll need to replace the mic cable with one that has the shell properly floated, or use a short XLR-female-to-XLR-male adapter with the shield disconnected from the shell. You won’t believe how much grief that can save you.
In the Channel
Once you optimize the levels coming from the various sound sources, you are ready to connect them to individual channel-strip inputs on your mixer.
Nearly every mixing console has a trim (or gain) control on each channel strip, and many consoles also include a pad switch, most of the time labeled as pad.
However, on many Allen & Heath consoles the mic/line selector switch is also used to engage the 20-dB pad for XLR sources. If in doubt, download and read the operator’s manual for your mixing console.
In any case, the pad and gain controls are used individually or in combination to make the level of the signal source coming into the console compatible with the input level of the channel preamp.
Mixer-channel pads generally reduce the input signal strength by a fixed amount, usually around 20 dB.
The pad is placed ahead of any transformers or other electronics in the circuit and should be engaged only when the input signal is too hot to be comfortably handled by the channel preamp.
The gain (or trim) control is a continuously variable potentiometer that adjusts the channel preamp gain. Microphone preamps typically offer up to 60 dB of gain boost, far more than most other gain stages in the signal chain, so be particularly careful when adjusting them.
If the input level is set too high, the preamp will be driven into clipping, causing distortion; if it is set too low, excessive noise will result.
Most consoles with a Solo function on the channel strip show the input level of a single channel on a meter when the channel is placed in Solo mode. To adjust the trim, zero all the controls on that channel strip and lower the fader completely.
Put that channel into Solo or Cue mode (typically called PFL for Pre Fader Listen on British consoles) and monitor it with headphones so you can evaluate the sound source for distortion or hum.
Have the vocalist or instrumentalist sing, talk or play his or her instrument while you watch the solo meter level; bring up the gain level until the meter approaches 0 dB on the loudest transients.
If you hear distortion on the headphones or the gain control can’t be turned down low enough to get the solo meter down to 0 dB on the peaks, engage the pad and bring up the input gain as appropriate.
In practice, you’ll probably want to set the channel level to peak somewhere between -6 dB and -10 dB on the PFL meter during sound check. Things tend to get louder during the actual show, and it’s preferable to incur a little noise rather than clipping the input stages when the sounds get heavier onstage.
Only after you have set the input level properly should you bring up the fader and add the signal source to the house mix.
Input gain levels can also change during the course of the show — guitarists are notorious for hedging their bets by playing softly during sound check and then cranking up the sound when the crowd arrives. Have no fear, though: if the signal level starts creeping up into the hot zone, you’ll probably notice the peak-overload LED on the channel strip blinking at you.
Similarly, you may find that you need to pull the fader down really low to make the signal fit in the mix.
In either case, adjust the input gain back on the offending channel and readjust the level of the channel fader in the final mix.
However, be aware that adjusting the input gain during a live show will also affect the monitor sends from that channel, which can make the musicians onstage very unhappy.
You may have to turn down the input while turning up the monitor sends to counteract the reduction in signal strength.
Still, that beats having a clipping channel sound bad for the entire service or performance.
Inserts and Loops
Once the source sound has passed the channel preamp stage, it is routed to the mixer’s internal mixing busses, but it can make several stops along the way.
Many mixers feature channel inserts that let you patch an outboard processor, usually a compressor or gate, into the signal path just after the preamp stage so that the entire audio signal must pass through it before reaching the EQ section and other internal circuitry.
Most channel inserts do not have send and receive level controls, so you will have to rely on the processor’s input and output level controls (assuming there are any) to set the gain at that point in the signal chain.
When using a compressor, the idea is to adjust the compressor’s output (or makeup) gain to compensate for any gain reduction caused by the compression process itself.
For example, if you read 10 dB of gain reduction on the compressor’s gain reduction meter, you may have to crank up its output level up by 10 dB to get the volume level back in the game.
Similarly, if you want to patch in an external processor without sending the entire signal through it (such as for reverb or echo effects), or if you want to be able to route signals to it from more than one input channel, the usual method is to connect the processor to an effects or auxiliary bus.
The signal from the effects or aux bus send is routed to the external processor’s input, and the signal from the external processor’s output is returned to the effects or auxiliary returns in the stereo output bus (or another mixer channel input if you want ultimate control).
Unlike channel inserts, effects and aux sends and returns nearly always have level controls.
Try setting the effects or aux send and receive levels in the mixer’s master section to their halfway points, and then slowly turn up the processor’s input level control until you get a consistently robust level. If the processor has a mix control, set it for 100 percent wet/effect.
Something else to consider when using external audio processors is their operating level. The insert points and effects busses on large professional mixers generally operate at a +4 dBu level, whereas those on lesser-grade mixers generally operate at -10 dBV.
Fortunately, many outboard processors can be switched between the levels. Look for a little button near the input jacks on the back that’s marked -10/+4 or something similar and set it accordingly.
If the processor’s input gain control must be set very low to prevent clipping the meter, you’re probably asking its -10 dBV input to handle a +4 dBu signal from the console, which is not nice to do.
In that case, trim back the input of the console strip itself until you can get the processor’s input control somewhere up around 50 percent.
Conversely, setting the processor’s input to +4 dBu for a console with a -10 dBV level will result in extra noise or not enough signal to drive the processor properly.
In critical listening situations you can also get transformer-based audio level shifters from companies like Ebtech or Whirlwind, which will boost or attenuate the levels from -10 dBV to +4 dBu or + 4 dBu to -10 dBV appropriately.
It’s important to think it all out in advance and listen during sound check to avoid bad audio during the actual worship service.
Finally, outboard processors can behave in dramatically different ways, so you need to understand each one.
For example, the overload LED on one processor might flash when the input signal reaches 6 dB below clipping, whereas on another it might not flash until the unit has been driven into distortion.
Or you might get a perfectly clean signal when cranking the output level of one processor to maximum and find that another one gets increasingly noisy past the halfway point.
Listen, then adjust, listen, adjust, etc…
On the Bus
The internal bus structure of a mixing console is also subject to headroom and S/N Ratio considerations. Whereas some consoles like to have their mixing busses driven hard, others’ buses can be clipped quite easily.
A good example of an inexpensive live console that needed its busses driven hard is an old Peavey console I had 30+ years ago.
There was a lot of noise in the mixing buses, but by running the output faders down around 2 or 3 (on a scale of 1 to 10) and driving the input stages a little hotter, it was possible to get a decent S/N at the outputs.
On the other hand, a more recent vintage Alesis 16-channel live console I used as a keyboard mixer didn’t have extra headroom in the mix bus but was very quiet. In that case, I ran the output faders up around 8 or 9 and then trimmed back the channel inputs until the output was at the right level.
The easiest way to determine the correct internal bus gain-staging approach is to plug in a dynamically consistent signal source, such as a drum machine or sampler, and listen with headphones for any crunching or distortion at the console output.
If there’s a lot of noise on the outputs with the faders up and no input signal present, bring the fader down until the noise is manageable.
Headphones make this easier to judge in a noisy room, so get yourself a quality pair and make friends with them.
If you hear distortion on the console outputs even when the meters read below 0 dB and the output faders are below halfway on the console, it means the internal mixing buses are clipping.
In that case, bring the input faders down and the output faders back up. High-end consoles have extremely quiet buses and a lot of headroom, so you typically won’t run into that sort of problem with them.
But many inexpensive consoles can be tweaked in the way I described to sound better than you might imagine. If you want to go further, you can use an oscilloscope and a signal generator to actually see flattening of the waveform and clipping in the various stages and adjust the levels accordingly.
Yes, it’s the ultimate geek thing to do, but oscilloscopes can be great troubleshooting tools.
Hit ‘Em Hard
You can also tweak the gain structure between the equalizer and the amplifier to improve the S/N ratio of the entire sound system.
For example, if you have sufficient gain from the equalizer’s output, you can raise its level by 10 dB and trim the input on the amp down by the same amount to attenuate any hum or ground-loop problems between the console and power amplifiers. That can really help in a quiet ambience mixing situation such as a church service.
Proper grounding, balanced inputs and shielded cables should, in theory, allow for an ultra-quiet connection between the console equalizer and the amplifiers.
However, that’s rarely the case in the real world. I’m always tweaking things one way or another to get the outputs as hot as possible without clipping and then turning down the inputs on the next stages.
End Game
Nearly any sound tech can properly operate a really expensive console with plenty of headroom and low noise, but it takes someone with real skills to make a cut-rate, unforgiving board sound great.
I have observed many guest engineers working with the same equipment get results ranging from fabulous to mediocre or worse, depending on how they ran the levels.
So don’t feel put down as a sound tech when you’re given some inexpensive gear and asked to make it sound great. Making an inexpensive system sound like a million bucks is the ultimate challenge. You can indeed spin straw into gold if you use your brains and experience.
Getting your church sound system to sound its best takes more than a great set of mixing ears for a particular music style.
It requires understanding how each piece of gear in the signal chain works and exploiting its potential to the max while working around any weak points.
Once you reach that level of knowledge, you are truly sympatico with the sound system and can make it do most anything you want.
Mike Sokol is the chief instructor of the HOW-TO Church Sound Workshops. He has 40 years of experience as a sound engineer, musician and author. Mike works with HOW-TO Sound Workshop Managing Partner Hector La Torre on the national, 36-city, annual HOW-TO Church Sound Workshop tour. Find out more here.
Return to article