PLAYS WELL WITH OTHERS: Staying out of the Red

In a perfect world, every device in your studio racks would integrate seamlessly with its neighbors, your levels would always be hot without a hint of

In a perfect world, every device in your studio racks wouldintegrate seamlessly with its neighbors, your levels would always behot without a hint of distortion, and routing signals would be likedriving down a smooth road on a sunny day. If that ever happened to me,I'd know for sure that I was about to awaken and find myself facedownon the mixer, drooling into the faders.

At my studio, I have collected some of the least expensive and mostincompatible audio gear ever made. Connecting that equipment is justthe beginning of a never-ending set of challenges. Once the signalsstart flowing through those rackmounted mongrels, it can be a long,bumpy ride down the treacherous road of gain staging.

In any integrated recording and mixing studio, includinguncomplicated cassette 4-tracks and self-contained digital-audioworkstations (DAWs), maintaining optimal gain at each stage of thesignal chain is crucial to the ultimate goal of clean, undistortedrecordings. (Any point in the chain at which the signal level can bechanged is called a gain stage.) If your signal dips too low at anystage, noise becomes part of the audio. In the analog domain, thatnoise — whether it's high-end hiss, radio-frequency interference(RFI), 60 Hz ground-loop buzz, or some combination of the three —is often amplified at one or more stages, especially when the audio isbeing compressed or equalized. (You should strive to minimize oreliminate those noise sources in any event.)

Digital quantization noise behaves in much the same way as analognoise, but it's harder to eliminate or filter out. The loss ofresolution that results from low digital-recording levels can never berecovered or corrected. At the other extreme, adding too much gain atany step of an analog or digital signal path introduces irreversibleand harsh-sounding clipping distortion (see Fig. 1).

Because a variety of reference standards are used in vintage,semipro, professional, and software-based audio gear, keeping yoursignal levels consistent throughout a studio-processing chain is alittle like driving on a rocky mountain road. When you route the audiofrom one device to the next, it can be a smooth turn, or you might feela little bump in the level. If you adjust the gain too much (or if gearset at a different operating level adjusts it without asking youfirst), you risk a grisly crash.


Fortunately, that motoring scenario includes a centerline designedto keep you from running into a mountain or careening off the side.Within the analog domain, where most audio signals originate, look fora zero mark — either 0 VU on a VU meter or 0 dB on an LED meter,fader, or knob (see Fig. 2). If you steer the level towardzero (allowing for normal fluctuations in dynamic level), your signalwill have a safe ride to its final destination.

The zero calibration of a device is referenced to its internaloperating level. A sine-wave signal that produces a zero-level readingon a compressor's output might show up as a different level on themeter of a multitrack tape machine to which it is routed. Patch thatsignal from tape to a DAW, and you'll get yet another meterreading.

Theoretically, a zero reading on a device's meter indicates aspecific AC voltage within a unit, calibrated to one of two prevailingaudio standards. The professional audio standard is +4 dBu. An analogdevice calibrated to that level references its 0 dB meter reading to aninput/output voltage of 1.23 VRMS, and the connection is generallybalanced. Stereo equipment, and some home-recording gear, is referencedto the consumer-level standard known as -10 dBV, in which a 0 dB meterreading indicates an input/output voltage of 0.316 VRMS and theconnection is typically unbalanced. (See “Square One: DecibelsDemystified, Part 2” in the August 2001 issue for more aboutthose standards.) When interfacing both kinds of equipment, it'simportant to know there is an 11.8 dB voltage discrepancy between -10dBV and +4 dBu reference levels.

Tape-machine inputs are referenced to -10 dBV or +4 dBu (someinclude inputs for both), but their meters are commonly adjusted sothat 0 VU actually indicates a level that is 3, 6, or even 8 dB higherthan the reference level. That convention lets engineers print maximumlevels on tape without continually pegging the meter. Film, video, andbroadcast equipment, especially older types, may also have uniquereference level requirements and other quirks.

If the devices in your studio are the same type (-10 dBV or +4 dBu)and each one is properly adjusted and performing no gain changes, a 0dB test tone applied to the first unit's input should appear at thefinal stage as a 0 dB level (see Fig. 3). But remember, I'mspeaking in theory. In the real world, components age or aren'tprecisely calibrated to begin with, and somewhere in your chain isprobably a box of the other type that you can't do without.


The initial source device, which is the first and most importantgain stage, also has the greatest impact on the overall signal-to-noiseratio. A careful level adjustment there — well above the noisefloor yet conservatively below clipping — will result in a clean,healthy signal throughout the rest of the chain as long as youintelligently use devices with different levels and observe theprinciple of unity gain.

Unity gain is a simple concept. When the input and outputlevels in a device are equal, resulting in no net gain or loss(technically, a gain ratio of one), that unit is operating at unitygain. The zero mark on a mixer's fader indicates unity gain between thecorresponding circuit's input and output. Running gear at or near unitygain preserves the optimal signal-to-noise ratio of the source anddownstream devices.

If your studio is set up and calibrated for one operating level (-10dBV or +4 dBu) and you follow the principle of unity gain (even withheavy-duty processing devices such as compressors and equalizers),you'll have few problems establishing proper gain staging. However, thereality is that most personal studios, and some world-class facilitiesas well, use a combination of -10 dBV and +4 dBu equipment. Thatmismatch is where most analog gain-staging woes arise.

ADAT, DTRS (the Tascam DA-88 and its siblings), and DAT machinesusually offer -10 dBV and +4 dBu input options with dedicatedconnectors to eliminate confusion (see Fig. 4). Connectortypes aren't strictly standardized, but RCA jacks typically carryunbalanced -10 dBV analog signals, and 3-pin XLRs or proprietarymultipin connectors commonly carry balanced +4 dBu signals.Quarter-inch connectors (TS or TRS) are often used in the shadowynether region, where mixed levels with balanced or unbalanced linescoexist. Unlike their analog cousins, no level differences existbetween the two common digital connections: S/PDIF (RCA jack) andAES/EBU (XLR connector).

In my experience as a mixed-level signal herder, it's easiest(though not essential) to keep signals consistent at -10 dBV or +4 dButhroughout a chain. My main multitrack recorder — a Tascam MS-16,16-track analog tape deck — offers -10 dBV and +4 dBu ins andouts. My mixing board, a Soundcraft Spirit, is set up to run at -10dBV, with mix bus outputs referenced to +4 dBu. (The deck that I usefor mixdown or mastering has +4 dBu inputs.) Accordingly, my tapereturns, board, and reverbs normally run at -10 dBV. The mixer'schannel inserts are also -10 dBV, and consequently, a lot of the rackgear usually used for mixing operates at -10 dBV.

However, I also have many mic preamps and compressors that I like torun at +4 dBu directly into the tape machine, bypassing the board andany extraneous connections. In situations like that, gear withswitchable -10 dBV or +4 dBu levels is obviously a godsend, and I willgladly take a trip to the back of my rack to change input or outputlevel switches when necessary. Nonetheless, combining mixed-level gearin a chain is certainly not out of the question and can still result inpristine audio quality. One particularly handy device in that regard isthe Ebtech Line Level Shifter (, which converts -10 dBV to +4 dBuand vice versa (see Fig. 5).


I rarely worry about decreasing a signal from a +4 dBu output to a-10 dBV input. Remember that zero on a -10 dBV unit is about 12 dB lessthan zero on +4 dBu equipment. If that arrangement produces a distortedsignal, attenuate the output level of the +4 dBu device by 10 or 12 dB;that should solve the problem with minimal impact on the audioquality.

However, I'm less carefree about stepping up signals from -10 dBVdevices to those operating at +4 dBu. When that gain change isunavoidable (for instance, from a -10 dBV keyboard to a +4 dBucompressor, or mastering from a consumer-level recorder to a pro deck),I step up the signal in one of two ways: boosting the internalamplifier of the source device or patching in another unit just foramplification.

Boosting the internal amp of a -10 dBV device by 12 dB is generallynot the best way to go; it can substantially raise the noise level andthe signal. In addition, the device may not have enough internalamplification to reach the required level. Finally, overdriving the -10dBV device's output amplifier beyond its available headroom results inclipping. When that happens, a meter on the downstream +4 dBu unit'sinput stage will read at or below zero and still sound distortedbecause the -10 dBV unit doesn't have enough output voltage to drive itat an optimal level. The far better approach is to use an externalamplifier to boost the level of the -10 dBV device by 12 dB.

Returning to my Soundcraft board for a moment, inquiring minds mightask, “If the board has -10 dBV inputs, inserts, and effects sendsand +4 dBu mix outputs, isn't there a step up involved?” Yes, butin that case, the console's high-quality amplifiers provide a gainboost with little extra noise. Another factor is that the signalscoming from my tape machine, referenced to -10 dBV, can easily havepeak values from 10 to 20 dB above zero. That brings the general signallevel at the faders within range of 0 VU at +4 dBu.

Once you become adept at analog gain setting, cautiously raiselevels through your system to take advantage of the various devices'headroom. When you're ready to do that, though, remember that theweakest gain link in the chain (invariably a -10 dBV device) will limitthe system's headroom. When you advance to that point, you will thinkof zero calibration as a guideline, but never an absolute or universalvalue.


Setting the level controls of all the devices in a signal chain is abalancing act; for example, if you increase the level at one point inthe chain, you must decrease the level somewhere downstream to maintaina constant average signal level throughout the chain. (The opposite isalso true: if you decrease the level at one point, you must increasethe level somewhere downstream to maintain a given signal level.)

In general, try to avoid setting any input- or output-level controlto its maximum value, which causes the circuit to operate at theextreme of its range and invites distortion and increased noise levels.Keep most gain controls somewhere in the middle of their range, whichlets the circuits operate in their most linear region, minimizing thepotential for clipping distortion and maximizing the signal-to-noiseratio. (That rule has some exceptions.)

Typically, the first gain stage in the chain is a mic preamp or anelectronic source, such as a synth, sampler, or CD player. Mostelectronic musical instruments don't offer output metering, nor do somemic preamps (even fully professional ones), which makes it difficult toset their average output levels to 0 dB. In that case, follow thesignal flow to the next downstream device — typically acompressor, recorder, or mixer — and set the initial gainaccording to what those meters tell you.

If you route the output of the source device directly to a tape deckthat has been calibrated to accurately reflect signal levels on itsmeters, simply use those meters to set the output level of the source.It's important to match the reference levels of both devices (that is,-10 dBV to -10 dBV or +4 dBu to +4 dBu). If the reference level of thesource does not match that of the deck, take steps to correct themismatch.

On the other hand, if you send the output from the source intoanother device with the capability to change the level, such as acompressor or mixer, that device's meter might not give you a clearpicture of what's going on. For example, if the input level of thedownstream device is set very low, its meter might show a level wellbelow zero, even if you crank the output level of the source to itsmaximum, which increases the potential for clipping and noise. Theproblem is compounded if you try to bring the meter reading up byboosting the output level of the downstream device to its maximum.

That is especially important in a mixer, where there are severalgain stages before the signal reaches the output meters (inputgain/trim, input fader, subgroup output, main output). If the mixerincludes a meter bridge with input-level meters, you still have tothink about the effect of the input gain/trim and fader on the meterreading.

Most +4 dBu devices have ample headroom to handle signal peaks above+20 dBu, but you can often hear clipping distortion at lower levels.The most common reason for audible clipping at levels that appearacceptable is inaccurate metering. VU meters are particularly slow toreact to transients, which makes them unreliable for gaugingfast-rising peak-voltage levels. Peak LED meters are more reliable inthat regard, and they can be calibrated to indicate clipping, levelsabove clipping, or a “safety zone” from 3 to 6 dB below theonset of clipping. If a manufacturer's specifications don't addressthat issue (and even in the rare cases when they do), setting thelevels according to peak-level meter readings involves trial anderror.

Another factor that complicates level setting by the meters isasymmetrical waveforms. Speakers, drums, and other common acousticsound sources tend to move equally far in both directions as theyvibrate, generating electrical waveforms that have equivalent positiveand negative voltage values. Those waveforms look symmetrical whenviewed on an oscilloscope or DAW screen.

However, some vocalists and most brass instruments and saxophonesgenerate waveforms that are decidedly asymmetrical with far greaternegative voltages when viewed onscreen (see Fig. 6). Olderpeak meters designed to read only positive voltage may not accuratelyreflect the level of those signals, letting the wave's negative voltagecomponent (which may be double the indicated voltage) clip and produceaudible distortion. Some signals, such as drum transients, mask somedistortion, whereas the identical amount of distortion is easilyaudible in a piano or nylon-string guitar note.


When routing a microphone signal through a mixer, it is standardpractice to set the channel fader at 0 dB and send the signal from thedirect or subgroup outputs to the destination recorder. The directsignal path is simple, and the input gain/trim control is the only gainstage that needs to be adjusted.

In that scenario, gain through the mixer channel is commonlymonitored by engaging the channel's prefader level (PFL) or solobutton, which routes the selected channel to the monitor bus anddisplays the level on the main stereo meters. On most mixers, it isalso possible to route an individual channel to a subgroup forlevel-setting purposes, even if that signal is sent out of the boardthrough the direct output. Many modern studio mixers includesignal-present or peak-overload indicator lights on each channel to aidthe gain-setting process. In addition to using those indicators, checkthe level at the recorder for any track being recorded.

When you're mixing one or more channels to a subgroup output,gain-staging procedures become more complicated. First, it's importantto establish the gain through each input channel by setting the channelfader at 0 dB and adjusting the input gain/trim control to maintain ausable signal level as indicated by the board's meters. Once that'sdone, assign the channels to the appropriate subgroups; their fadersshould be set to 0 dB.

To create a mix of two or more channels in a subgroup, simply adjustthe channel faders and perhaps bring the overall subgroup level up ordown. Use the meters on the subgroup and the destination recorder todetermine the subgroup master level. Once the submix is set, all fadersshould ideally be within ±10 dB of zero. If not, adjust the inputgain/trim to bring the channel fader into the recommended range andmaintain an optimal signal-to-noise ratio.

Within a mixer are several other gain stages related to theauxiliary effects buses. Those controls are typically rotary pots thatmay have a central or unity-gain setting flanked by numerical plus andminus values, or they might use a simple one-to-ten numbering scheme.In addition to aux-send controls in each input channel, each aux busshould also have a master-send control and an after-fader level (AFL)or solo button that lets you check the output signal to make sure it'swithin a usable range (that is, an average level near 0 dB).

Most engineers maximize the headroom within outboard reverb units bysetting input and output controls to their full 100 percent levels. Inmost instances, they also set the mix controls (which adjust the ratioof dry to processed sound) to full or 100 percent “wet.”Most digital effects processors have their own metering with clearpeak-level indication. For best audio quality, send the hottestpossible signal to the units but avoid peaking and listen carefully fordistortion.

The final gain stage in the aux-effects chain is the effects returnat the board. That control can be a rotary pot or fader, and it adjustshow much effected signal is added to the stereo mix bus. In most cases,that effected signal is far less than a 0 dB level. Effects-returnlevels must be adjusted by ear rather than by metering. Solo or AFLswitches at the effects return let you determine if the processedsignal is undistorted and at a usable level. Comparing the effects-sendsignal to the return signal can also supply valuable insight into thecharacter of an effects program. In addition, such a comparison canindicate overall gain changes through the unit, which may be due toreference-level mismatches or extreme regeneration in delayeffects.

Compressors also present unique gain-setting challenges. As ageneral rule, you should compensate for the amount of gain reductionindicated on a compressor's meter by setting an equivalent boost at themakeup gain control. For example, if the compressor regularly cuts asignal by -4 dB, a makeup gain boost of +4 dB keeps the peaks atroughly the same level through the unit. The signal will sound louderbecause the dynamic range is reduced and low-level signals are boosted.But in terms of peak metering, the maximum signal levels should be thesame from input to output and more consistent as well. At extremecompression settings, that method may not be so predictable, and youshould watch the meters in a downstream device after thecompressor.


When it comes to setting digital levels, 0 dBFS (decibels fullscale) is an absolute ceiling value that should never be reached untilthe final stage of mastering. At that level, the highest peaks of thewaveform are represented by binary numbers consisting of all ones.Think of 0 dBFS as the guardrail on your digital gain-staging road;it's always there to guide you, but you sure don't want to risk runningup against it!

There are various philosophies about what levels should bemaintained when mixing from analog to digital or recording from ananalog board into a modular digital multitrack (MDM) or DAW. Like theproducers of analog gear, manufacturers of digital equipment usevarious digital-reference standards meant to keep engineers from maxingout their dBFS levels when making analog-to-digital transfers.

The conventional wisdom is that 12 dB of headroom is generallyacceptable in a digital-audio recording system. That means an averagemeter reading of 0 dB on an analog mixer's outputs should equate toabout -12 dBFS on a digital meter (assuming that the mixer's outputsand the MDM's analog inputs are at the same reference level). I say“about,” because no established reference-level standardexists within the industry for the crucial operation of convertinganalog audio voltage into digital ones and zeros.

Many equipment manufacturers reference 0 dB to -18 dBFS, whereasothers set their standards from -12 to -24 dBFS. To keep overzealousrecordists from pegging the meters (as they did in the old days ofanalog), some companies also include a safety margin — a fewdecibels of “hidden” headroom — between the 0 dBFSpeak indication and the point of digital clipping. That lets you scootalong the mythical guardrail without scraping up your paint job.

Many MDMs offer -10 dBV and +4 dBu inputs, and the smoothest ride isalways on the road where your reference levels match. In many studios,outboard mic preamps referenced to +4 dBu run straight into digitalrecorder inputs, following the philosophy of a minimal signal path.That routing presents few gain-staging problems as long as you rememberthat such a mic preamp needs to provide roughly 12 dB less output whendriving a -10 dBV input.

On rare occasions, I've found that a +4 dBu preamp running at itslowest output still provided too much level for a -10 dBV ADAT input,and it was not feasible to use the multipin +4 dBu Elco snake. In thosecircumstances, it is easiest to engage the pad switch (if one isprovided on the mic or preamp) or physically move the microphonefarther from the source. If that doesn't do the trick, the remainingoptions are to choose another mic or preamp with lower output level orto insert a compressor or other gain-reducing device between the preampand recorder.

There is also the possibility that a -10 dBV unit, such as akeyboard or CD player, may be called upon to drive a +4 dBu input on adigital device. An intermediate line amplifier, such as the Ebtech LineLevel Shifter or the internal amp in a +4 dBu mixer, is the best way toavoid step-up problems. The increase in resolution attained by keepingthe signal as hot as possible generally outweighs the potentialincrease in noise you may get by boosting levels in that manner,especially in a 16-bit digital-recording format.


Stereo mixdown to a digital medium such as DAT is another commonprocedure. When mixing from an analog console, remember that a signalreading 0 dB on the mixer's meter typically measures between -12 and-18 dBFS on the digital recorder's meter, depending on theanalog-to-digital converter's calibration. If your program material hasa wide dynamic range, set an average signal level of 0 dB at themixer's master faders (once you've established proper gain stagingthroughout the rest of the board) and then play the selection frombeginning to end, noting the highest peak level (not merely the averagelevel) registered on the digital recorder. The peak-hold function onmany digital meters is particularly helpful for logging maximumlevels.

After the level-checking pass, you may want to adjust the DAT'sinput level, keeping in mind that it's always good practice to keeppeak signal levels 2 to 3 dB below 0 dBFS to retain the bestresolution. If you find that one transient peak in your piece hits -2dBFS and the rest of the program sits around -8 dBFS, considercompressing or manually fading the track with the offending transient.That lets you boost the entire mix level by 6 dB and utilize moreavailable bits.

On average, a highly compressed rock mix might have only 4 to 8 dBof dynamic range. Transferring such a mix at 0 dB equal to -12 dBFS isa waste of perfectly good bits. For example, if the dynamic range ofthe mix is 8 dB, set the DAT's input level so that 0 dB (the averagelevel of the mix) corresponds to -6 dBFS, which lets the dynamic rangevary by ±4 dB while leaving 2 dB of headroom below 0 dBFS.

Whatever kind of mix you have, do not be afraid to nudge the levelso that you have a strong average digital level and no peak goes higherthan -2 or -3 dBFS. Don't forget to use your ears on the final product.Some digital meters are a bit slow; they can let a distorted transientslip by undetected if every sample of the incoming audio is notmeasured.

In an all-digital mix — whether from a digital mixer or withina self-contained recording and mixing DAW such as Pro Tools — payspecial attention to digital headroom. In a 16-bit system, it'sadvisable to keep peak levels between -2 and -3 dBFS. Higher-resolutionsystems can reproduce a greater dynamic range, and peaks near -6 dBFSare acceptable in that case. The internal processing of a digital mixermight allow for extra headroom and longer word lengths, but its signalmust ultimately pass through a digital-to-analog converter to be heard.Also, the ultimate destination is usually a 16-bit CD. Those factsoften mandate a gain reduction at the output faders to keep the finallevel below the 0 dBFS ceiling, especially with dense multitrackmixes.


The best mindset for proper analog gain staging is to be aware ofthe potential consequences of stepping the gain up or down inprofessional and consumer-level gear. Even if the analog portion ofyour studio is set up and precisely calibrated for only one referencelevel, you should still take the time to maintain and periodicallycheck zero levels wherever possible in the processing chain, followingthe signal flow's direction.

Within the digital domain, it's equally important to maximize yourlevels, but keep the peaks at -2 to -3 dBFS at all stages before finalmastering. It is also crucial to listen for distortion and noise at theend of an analog or digital signal chain, regardless of what the meterstell you.

To avoid nasty potholes and detours on the gain-staging road, steeryour signals toward zero, keep your eyes on the gauges, and drive withyour ears. Armed with the knowledge thus acquired, your signal-routingtrip can indeed be like driving down a smooth, level road on a sunnyday.

Myles Boisen ( is a guitarist, producer,composer, and head engineer and instructor at Guerrilla Recording andthe Headless Buddha Mastering Lab in Oakland, California. Thanks toKaren Stackpole, Bob Smith of BS Studios, and Lawrence Fellows-Mannionof Rance Electronics.


When setting up a studio system, start with the sources — inmost cases, synths or samplers with -10 dBV outputs and one or moremicrophones connected to mic preamps with +4 dBu outputs. To ensurethat the synths have adequate gain, connect their outputs to directinjection (DI) boxes and connect the DI boxes to mic preamps. (For micsand synths with DI boxes, it's best if the preamps have output-levelmeters.) If DI boxes are not available, connect the synths' outputs tothe -10 dBV inputs on the mixer.

In that procedure, you will use one of the sources as a signalgenerator to set the levels throughout the rest of the signal chain. Atest-tone generator with a known output level is more reliable for thatpurpose, but it's not pleasant to listen to for an extended period oftime. A sustained keyboard chord is fine, as is a microphone in frontof a radio or other compressed music source.

If you use a phantom-powered condenser mic, connect it to the preampand apply phantom power first. Then, connect and power up all studiodevices except for the final power amp or powered monitors. Set thegain on outboard devices to unity. Start with all mixer controls atminimum or detented settings; EQs should be flat or bypassed.

If you use an outboard preamp as the source, raise the gain so thepreamp meter reads 0 dB. (If you connected a synth to the preamp, setthe synth's output to the highest level that does not overload thepreamp's input.) Connect the preamp directly to each outboardcompressor, setting the controls for unity gain, no gain reduction, and0 dB on the output meter. Any other outboard insert devices used intracking (gates, equalizers, and so on) should also be calibrated inthat manner.

Once it's been calibrated, the final device in the chain can berouted to one channel of the multitrack recorder to check its meterreading. The preamp's 0 dB level might not read zero on a digital oranalog recorder's meter. For calibration or adjustment of the deck'sinput level, consult a qualified technician.

Next, route the preamp's output to an appropriate line input (notmic input) on the mixer. Set the mixer's channel, subgroup, and masterfaders to zero or unity; engage the channel's PFL or solo switch; andraise the channel's gain/trim control until the meter reads 0 dB.(Repeat for all other input channels.) At that point, the subgroups andmaster buses should read zero as well. If you're not using an outboardpreamp, connect the mic or synth to an appropriate input on the mixerand follow the same procedure.

Connect the channel-direct outputs or subgroup outs to themultitrack recorder and check the levels on the multitrack meters. (Asbefore, the mixer's 0 dB level might not read zero on the recorder'smeters. For calibration or adjustment of the recorder's input level,consult a qualified technician.) Connect the recorder's outputs to themixer's tape returns, engage the deck's monitor function, and adjustthe mixer's tape-return levels so that they indicate zero on themixer's meters after pressing the appropriate PFL or solo switch.

Connect the channel inserts to devices used for tracking or mixing.Generally, those devices are compressors, gates, equalizers, digitaldelays, and other single-channel effects not connected to an aux bus.Make sure the reference levels of those units match the mixer insert'soperating level. Check for unity gain on the PFL or solo meters or bymonitoring the channel level by ear while physically engaging anddisengaging the insert connector from the mixer.

Set the inputs and outputs of all aux effects units at maximum andset the mix control at 0 percent (completely dry, noneffected signal).On the active mixer channel, set the individual aux-send levels at arepeatable midway point (for example, unity, 5 o'clock, or 12 o'clock).Raise the master aux-send control until the effects unit indicates a 0dB level.

If a PFL or AFL meter is available to indicate the level at themixer's master effects return, check it to make sure that a usablesignal (that is, near 0 dB) is coming back into the board with themaster effects-return control set between the midway and maximumpoints. Then, monitor those levels with the effects unit's mix controlset to 100 percent. Various effects units will probably calibratesomewhat differently through those steps, but as long as levels are inthe usable range (near 0 dB) without distortion, that should not because for alarm. Note the positions of all effects-send and -returnmasters so they can be preset before a mix.

Next, route the active channel to the 2-track mixdown recorder (DAT,CD, hard disk, or analog) and adjust its input levels so the deck'smeters read zero when the mixer's stereo output bus reads zero. (Anoutboard sampler can be calibrated the same way.) To establish unitygain through the mixdown deck, connect the deck's outputs to themixer's 2-track inputs (or other inputs if it has no dedicated inputs),record the calibration signal you're using, and then play it back andadjust the deck's output until the mixer's meters read 0. Make sure thereference level of the 2-track or sampler matches the mixer's outputand 2-track input reference level (-10 dBV or +4 dBu).

During those procedures, it is possible (though not necessary) tomonitor the audio signal with headphones, which can also be used tocheck the gain in any headphone mix buses following the proceduresoutlined for the aux buses. In addition, adjust and note comfortablelevels through headphone distribution amps. Bear in mind that differentheadphones have widely varying efficiency ratings, and one brand ormodel may be much louder than another.

For the final stage, make sure your mixer's stereo master andcontrol-room output faders are all the way down. Then, power up yourmain monitor amplifier(s). If the amp has gain controls, set them at100 percent. Raise the mixer's main faders to zero and slowly increasethe control-room level until a comfortable listening level isestablished. If a sound pressure level (SPL) meter is available, setthe listening level between 80 and 90 dB. Clearly mark the position ofthe control-room master for future reference as a safe setting.

If the safe control-room setting seems low (say, below three on apot marked from one to ten) or the monitors are noticeably noisy, youmight want to attenuate the amp gain. If you do that, turn down bothsides of a stereo amplifier by the same amount and test for audiblestereo balance using a high-frequency mono source panned to center. Inaddition, be aware that trimming a power amp's gain lowers itsheadroom, which may interfere with transient response. Clippingdistortion may also occur if the mixer has to overdrive the amp's inputto achieve the desired listening level. For those reasons, mostprofessionals run their amps at 100 percent and take care to keep thecontrol-room pot at a conservative setting.

Another approach...

by Scott Wilkinson

Another approach to gain staging also works well in the studio. Thegoal is to minimize the audible noise level while maximizing the signallevel at each gain stage. The procedure yields a wonderfully quietsound system with plenty of signal level.

The idea is to work backward from the control-room amp to the mixerand aux effects and, finally, to the sources. EQ should be completelyflat as you work through these steps.

  1. Connect everything.
  2. Set volume controls to minimum.
  3. Power up all devices in order: instruments and preamps, effects,mixer, amplifier.
  4. Raise mixer control-room volume to maximum; back off until noisedisappears.
  5. Raise amp level to maximum; back off until noise disappears.
  6. Raise mixer master level to maximum; back off until noise disappears(hopefully, unity gain).
  7. Raise each mixer subgroup fader to maximum; back off until noisedisappears (hopefully, unity gain).
    For each aux effect bus:
  8. Raise effects mix or wet/dry control to 100 percent or full wet.
  9. Raise effects input and output levels to 100 percent.
  10. Raise mixer effects-return level to maximum; back off until noisedisappears.
  11. Raise mixer master send level to maximum; back off until noisedisappears.
  12. Raise each channel aux-send level to maximum; back off until noisedisappears.
    Adjust as needed for each channel during mix but try not to exceed thatlevel.
    For each mixer input:
  13. Raise the mixer input fader to maximum.
  14. Raise level of instrument or mic preamp to maximum.
  15. Raise mixer input gain/trim to maximum; back off until noisedisappears.
  16. Lower input fader to unity gain.
  17. Lower instrument or preamp level to minimum before playing.
  18. Slowly raise instrument or preamp level to desired level whileplaying, as indicated on mixer or tape-deck meters.