TAKE IT TO THE LIMIT: Push your P.A. to peak performance

Jan 11, 2008 5:42 PM, By Mike Sokol

           

INSERTS AND LOOPS

Once the source sound has passed the channel preamp stage, it is routed to the mixer's internal mixing buses, but it can make several stops along the way. Many mixers feature channel inserts that let you patch an outboard processor, usually a dynamics processor, 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. For example, if you read 10 dB of gain reduction on the compressor's gain reduction meter, crank the output level up 10 dB.

Similarly, if you want to patch in an external processor without sending the entire signal through it, 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 routed to the effects or aux return (or another mixer channel input). 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 effect.

Something else to consider when using external audio processors is their operating level. The insert points and effects buses on large professional mixers generally operate at a +4 dB level, whereas those on semipro and consumer-grade mixers generally operate at -10 dB. Fortunately, many outboard processors can be switched between the levels.

If the processor's input gain control must be set very low to prevent clipping the meter, you're probably asking its -10 input to handle a +4 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 dB for a console with a -10 dB level will result in extra noise. It's important to think it all out and listen during sound check.

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, 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.

ON THE BUS

The internal bus structure of a mixing console is subject to headroom and S/N considerations as well. Whereas some consoles like to have their mixing buses driven hard, others' buses can be clipped quite easily. A good example of an inexpensive live console that needed the buses driven hard was my old Peavey console from 20 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, I had a more recent vintage Alesis 16-channel live console that didn't have extra headroom in the mix bus but was very quiet. In that case, I ran the output faders around 8 or 9 and then trimmed back the inputs until the output was at the right level.

The easiest way to determine the correct gain-staging approach is to plug in a dynamically consistent signal source, such as a drum machine, and listen for any crunching or distortion at the console output with headphones. If there's a lot of noise on the outputs with the faders up and no input signal present, bring the fader down until it's manageable.

If you hear distortion on the console outputs even when the meters read below 0 VU and the output faders are below halfway on the console, 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 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 clipping in the various stages and adjust the levels accordingly.

HIT 'EM HARD

You can also tweak the gain structure between the equalizer and the amplifier to improve the S/N. For example, if you have sufficient gain from the equalizer output, you can raise its level by 10 dB and trim the input on the amp by the same amount to attenuate any hum or ground-loop problems between the console and amplifiers. That can really help in a quiet mixing situation.

Proper grounding, balanced inputs, and shielded cables should, in theory, allow for an ultraquiet 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.

ENDGAME

Nearly any engineer can properly operate a really expensive console with plenty of headroom and low noise, but it takes an engineer with chops to make a cheap, 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 an engineer when you are given some inexpensive gear and asked to make it sound great. To me, making an inexpensive system sound like a million bucks is like the ultimate challenge to spin straw into gold.

Getting a P.A. 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, you are truly one with the P.A. system; you can make it do anything you want.

Mike Sokol is a live-sound and recording engineer, musician, and computer integrator with 30 years' experience on both sides of the microphone. He lives in western Maryland with his wife, four boys, and three cats.

onstage•hotlinks

http://otto.cmr.fsu.edu/~elec4mus/topics/decibel.html
An article by Glenn While at the Florida State University School of Music defines technical audio terms related to gain structuring.

http://prosoundweb.com/studyhall/studyindex.html
The Study Hall section of ProSoundWeb.com contains technical articles on clipping, audio math, gain structuring, and related topics.

www.eatel.net/~amptech/elecdisc/sig2nois.htm
A primer on signal-to-noise ratios has automated calculators.

Digital Matters

Digital gear's operating characteristics are different from analog gear's, and those differences need to be taken into account when structuring gain, particularly when digital and analog devices are interfaced in a signal chain. For example, the inputs on analog gear can typically be driven to +2 or +3 dB before any detectable distortion is introduced, whereas with digital gear, as soon as you pass 0 dB, you're likely to get nasty digital distortion of the most unforgiving kind.

At the same time, if you are using, say, a 16-bit digital reverb unit, one bit of its word length will be used for each 6 dB of audio. So if you keep the signal to the input so low that you never get as high as -12 decibels full scale (dBFS) on the input meter, you're working with only 14 bits. If you run the input in the -24 dBFS range, you're asking the processor in the reverb to work with 12-bit audio, which can sound quite grungy. So make sure that the input level is in the -10 dBFS range most of the time and that it never quite peaks out.

Another difference between digital and analog gear is that with digital gear you can usually crank the output level all the way up without getting any distortion or unwanted artifacts (though some engineers go only to 90 percent, just to be safe). With a robust input level and the output level set to full, you probably won't need to turn the mixer's effects return level up very high to add an appropriate amount of reverb into the mix. If you were to go the other way around and just barely drive the reverb's input while boosting the return to get enough reverb level, you'd most likely hear lots of quantization artifacts and other noise as a result.

These considerations apply when interfacing any digital and analog gear, so think everything through as you analyze your audio signal chain.