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The Path of Least Resistance - EMusician

The Path of Least Resistance

There is no question that direct recording with good gear will give you the cleanest sound. And no instrument reflects the efficiency of direct recording
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There is no question that direct recording with good gear will giveyou the cleanest sound. And no instrument reflects the efficiency ofdirect recording more than the human voice. However, the average vocalsignal takes a trip through various gain stages in the mixer's channelstrip, through the send/returns to effects (such as compression andEQ), and down the mix bus before hitting the recorder. Electricallyspeaking, that adds up to quite a distance and can translate into addednoise and signal degradation.

For avoiding such noise and degradation while still getting theeffects essential to recording vocals, there are handy devices on themarket that reduce the number of parts the signal goes through and thusminimize the length of the path from input to output. These devices areoften referred to as channel strips and voice processors.

ONEVOICE, ONE CHANNEL The terms channel strip and voice processor areoften used interchangeably. That's because both kinds of units have asimilar purpose: to get the signal to the recorder as directly aspossible. But the two types also have significant differences.

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Stand-alone channel strips are designed to take the place of amixer's channel strip. The stand-alone has the same features as thechannel strip on a mixer: mic preamp, line-level input, shelvingfilters, and parametric EQ. Besides the shorter signal path, the bigdifference is that the stand-alone version contains better componentsthan those in the average personal-studio mixer. When you spend $600per channel, you get components comparable to those in a high-endmixing desk.

Voice processors are also designed for direct recording and containthe same features as a channel strip, but with a few additions.Engineers regularly use dynamics processors when recording vocalsbecause of the voice's wide dynamic range. That means you will want acompressor, limiter, expander, gate, and de-esser in the unit. Ofcourse, these effects work well on other instruments, too, such aselectric guitar, bass, and keyboards.

The layout of a voice processor should be as intuitive as that of achannel strip. In a mixer, the signal generally flows from top tobottom. The signal flow in a voice processor is similar, except thatyou lay the unit on its side so that the signal runs from left toright, entering the input stage at the far left, going through thevarious effects stages, and ending up at the output on the farright.

Don't limit yourself by using voice processors only for tracking.These units can prove equally beneficial during mixdown or forfattening prerecorded tracks. For this reason, voice processors have avariety of input and output options to maximize their usefulness in thestudio.

CHOICES AND MORE CHOICES Manufacturers all have different ideas asto what their direct-recording device should include and how thefeatures should be organized. Because of the enormous number ofpossibilities, I have limited myself to discussing a handful ofproducts for the purpose of comparison.

Voice processors can range in price from around $500 to $4,000. Idecided to look at single-channel units priced between $630 and $850that combine a mic preamp, compressor/limiter, expander/ gate, and EQ.Within these parameters, I looked at a host of other useful features(for example, tube emulation, digital converters, and independentlyaccessible effects), and chose a collection of processors thatcomplemented, rather than fully duplicated, each other.

Besides the five that I have chosen to cover here, there are manyother voice processors in this $630-to- $850 range. Therefore, as Idiscuss specific features, I will also briefly point to some of theseothers as examples. But before we examine our five contenders, let'sreview the various features that make up a voice processor.

INS AND OUTS When you're looking for the ideal direct-recordingsolution that is also useful for mixing, the more I/O possibilities youhave, the better. In addition to a mic input, there should be line- andinstrument-level inputs. The ability to tap in and out of theindividual effects is a plus. At the other end, having balanced (+4dBV, 11/44-inch TRS or XLR) and unbalanced (-10 dBu, 11/44-inch TS)line-level outputs is essential. Let's take a closer look at eachsection.

Mic preamp. The microphone preamp is one of the most important partsof a voice processor because it sets the stage (literally) for yoursound. In order to record with the best signal-to-noise ratio possible,the output level of the microphone (-30 to -60 dBu) should be broughtup to line level (-10 dBu or +4 dBV).

Usually, this change in level will color the sound somewhat, and inmany cases that's good. One joy of engineering is tone sculpting, andthe mic preamp can have as much to do with this sculpting as themicrophone itself. For example, a tube preamp will have a differentcharacter than a solid-state unit. In addition, Class A circuitry(solid-state or tube) will give you superior sound and performance butwill often cost a little extra.

Ultimately, the degree of transparency or coloration that you wantthe preamp to impart depends on your personal taste and the demands ofthe music. Other considerations include the recording medium thatyou're using. Engineers with tapeless studios may want a preamp thatadds "tube warmth" or can emulate the effects of tape saturation. Onthe other hand, users of analog tape may want a transparent sound, withas little coloration as possible. The call is a completely subjectiveone. What sounds harsh and buzzy to one engineer may sound warm andfuzzy to another.

Another consideration is how your collection of mics will soundthrough the preamp: invariably, some mics will sound better thanothers. There are no rules that say you can't run a cheap mic throughan expensive preamp or vice versa-which preamp you choose should bedetermined by whatever works best for the style of music that you'rerecording.

Examples of voice processors with tube preamps include A.R.T.'s ProChannel ($799), Manley's Voxbox ($4,000), and Avalon Design's VT-737SP($2,295). Solid-state designs include the Focusrite Voicebox MKII($1,345) and the Rane VP12 ($599). The Manley, Avalon, and Focusriteare more expensive because they use Class A circuitry.

A mic preamp can include a couple of features that make life easier.First, it should have +48 volt phantom power so that you can power themics that need it. A phase-reversal switch is also useful for changingthe phase of the signal by 180 degrees. In addition, preamps often havea -10 or -20 dB pad to attenuate high- output mics and a low-cut filterto remove low-frequency energy such as rumble. Most voice processors onthe market include many or all of these features.

Instrument input/DI. If you have ever recorded electric guitar orbass, you know how handy it is to have a compressor, gate, and EQ atyour disposal. Having access to a tube stage is another plus. You'llprobably want to plug your instrument into your voice processor andtake advantage of these effects. Fortunately, many of these units havea 11/44-inch, instrument-level input jack. The instrument input (or DI)raises the signal from instrument level (-30 to -20 dB) to line level.Look for processors with convenient 11/44-inch front-panel jacks.

Line-level input and output. In addition to using a voice processoron mics and guitars, you can use it on keyboards, sound modules, andprerecorded tracks during mixdown. Manufacturers often provide both +4dBV balanced and -10 dBu unbalanced line-level inputs to facilitatethis type of use.

Voice processors also typically have a -10 dBu and a +4 dBV output.Depending on your studio situation, having the option of both outputlevels means that the device will have upward compatibility if you'removing from so-called semipro gear (which typically operates at -10) toprofessional gear (operating at +4).

Compressor/limiter. The dynamic range of the voice goes beyond whatthe average recorder can handle, so engineers routinely use compressionwhen tracking vocals. The type of compressor and how well it operatescan impact how the signal sounds after the fact. Consequently, takegreat care when choosing and using a compressor.

With a dedicated compressor you will usually have control overthreshold level, attack and release times, ratio selection, and make-upgain, as well as the choice between hard or soft knee (that is, howsuddenly compression begins).

Limiting is at the extreme end of compression, when the ratio is10:1 and beyond. Limiting effectively inhibits any increase in outputlevel no matter how far the signal goes above the threshold.

Compressors and limiters have three common designs: photo-optical,VCA, and variable mu. Photo-optical compressors (also calledphotoelectric or just plain "opto") differ in several ways from thosecompressors governed by a voltage-controlled amplifier (VCA). Optocompressors use a photosensitive resistor controlled by light-emittingdiodes that are triggered by the input signal. Opto units compress overa narrower range (with less extreme compression ratios) and impart alivelier sound to the signal. Because of their simple user interface,they lend themselves to tweaking by ear. Examples of photo-opticalcompressors include A.R.T.'s Tube Channel ($499), Avalon Design'sVT-737SP, and Focusrite's Platinum VoiceMaster ($749).

VCA units, on the other hand, are far more predictable, are easierto use, have greater ratios, and allow for more flexibility andprecision with attack and release times. Voice processors with VCAcompressors include the MindPrint En-Voice ($749), PreSonus VXP ($700),and Focusrite Red 7 ($2,995). The En- Voice compressor includes a tubesaturation stage.

Variable mu compressors (mu in this case means "gain") use a vacuumtube as a variable resistor to control the signal level. Manley uses acombination of variable mu and optical compression/limiting in itsVoxbox. The A.R.T. Pro Channel gives you the choice of either opto/tubeor variable-mu gain control.

Expanders and gates. Downward expansion and gating are processesused to reduce noise. Expanders create a greater dynamic range byprogressively attenuating sounds that fall below a specified threshold;that is, they make the quiet sounds quieter. You can think of anexpander as the opposite of a compressor, which brings up the level ofthe quieter sounds.

A full-featured expander gives you control over the threshold wherethe processing begins, the ratio of expansion, and the speed of theprocessing. Expanders are used primarily for eliminating extraneoussounds such as bleed from headphones, lip smacks, and clothing noises.Usually used during mixing rather than recording, the expander is oftenplaced after the compressor to reduce the low-level noise thatcompression brings out. Many voice processors, however, locate theexpander before the compressor, giving you the opportunity to reducenoise before the compression stage.

At the extreme end of downward expansion is gating. Rather thansimply reducing the signal level when it drops below the threshold, agate shuts the level off completely. Gating is used as much to shapethe envelope of signals as to remove unwanted noise. For example, gatesare used to fatten up drum sounds by quickly attenuating theinstrument's natural decay. You can also use a gate to keep the buzz ofan amp from passing through the unit when a guitarist isn't playing.Typical gate parameters include threshold and rate.

EQ and filters. Direct control over the timbral aspects of a signalis always welcome. Whether you use EQ to enhance the signal, removeunwanted anomalies, or repair damage caused by compression, you willwant a band or two of parametric EQ as well as high- and low-shelvingfilters.

A parametric EQ lets you control three essential parameters: theexact frequency that you want to affect; the bandwidth, or Q, which isthe range of frequencies around the center frequency; and volume forcutting or boosting.

Shelving filters allow you to modify the extreme ends of the signaland are usually found in the input section of voice processors. Themost common is the highpass filter (also called a low-cut filter) whichattenuates everything below a specific frequency. This is especiallyuseful for removing rumble and mic- handling noise. Because the humanvoice doesn't go below 80 Hz, a typical cutoff frequency for a highpassfilter is 75 Hz. Some units give you variable control over the cutfrequency, sometimes ranging from 15 to 320 Hz. At the other end of thespectrum is the lowpass (or high-cut) filter, which affects everythingabove a specific high frequency, such as hiss.

For those of us with a limited number of mics, parametric EQsupplies the tools with which to expand the sonic palette. EQ is also amust when tracking instruments such as the electric guitar. Ranges inwhich the guitar can often use help include the lower-mid range, around500 Hz; between 3 to 6 kHz for added bite; and from 8 to 10 kHz forsparkle. Unlike a graphic EQ, a parametric EQ enables you to locate theexact frequencies that need attention.

De-esser. A de-esser provides a form of frequency-dependentcompression used to remove sibilance when recording vocals. Sibilants,such as s and sh sounds, have a high-frequency concentration between 3and 8 kHz. Traditional de-essing is done by splitting the vocal signal,sending one side through the compressor's audio input, and the otherside to an EQ patched into the compressor's sidechain input. The exactsibilant frequencies are boosted on the EQ, which makes the compressormore sensitive to them. The EQ-enhanced signal causes the compressor toattenuate the sibilants in the direct signal. Meanwhile, the signalgoing into the sidechain is not heard.

A de-esser increases the usability of a voice processor. However, apoorly implemented de-esser can sound more like a lowpass filter than afrequency-dependent compressor.

Insert and sidechain. As with a mixing console, an insert lets youintroduce an additional outboard processor into the signal path. Italso gives you an extra output point before the main output stage.

The sidechain input, a common item on dedicated dynamics processors,lets you control the compressor with an external signal. This is usefulfor ducking and frequency-dependent compression.

THE CONTENDERS In designing a product such as a voice processor,manufacturers make presumptions about how recording engineers like towork and the kinds of features that they need most. For the sake ofcontrast and comparison, I have selected five units that have as manyof the above features as possible, with just as manyimplementations.

dbx 1086. The dbx 1086 Mic Pre Processor ($750) stands out in thecrowd in several ways. The most interesting for the personal-studioowner is that the mic preamp and the dynamics processor can be usedindependently. This allows you to simultaneously track a vocal whileprocessing another signal. The other great feature is that the 1086 hasroom for an optional A/D converter, the dbx 504X. However, the 1086 isthe only unit of the group that lacks a true parametric EQ section (seeFig. 1). The 1086's preamp section has an XLR input and an XLR and11/44-inch TRS output. The dynamics section has XLR and 11/44-inch TRSline-level inputs and outputs. Both output sections have a switchallowing you to choose between -10 dBV and +4 dBu output levels (seeFig. 2).

The preamp section has a variable-frequency low-cut filter (30 to300 Hz) and a 2-band additive filter called Detail. When switched in,the Low control simultaneously boosts 125 Hz and cuts 400 Hz at a 2:1ratio. The High knob adds up to 15 dB of the "air band" above 10 kHz.Having a variable-frequency Low Cut next to the Detail section baffledme initially. But I could imagine a scenario in which you would want toboost 125 Hz while cutting 70 Hz at the same time, and this unit willgive you that option.

The 1086 has a single bypass switch for the entire dynamics section.However, the controls for the expander/gate, compressor/limiter, andde-esser each have an off position that effectively removes them fromthe signal chain. All of the buttons on the 1086 are backlit so thatyou can easily see their status, and the rotary controls use40-position stepped, rather than continuously variable, pots.

The expander/gate has variable threshold and ratio controls, thecompressor has threshold, ratio, and output-gain controls, and thede-esser lets you set threshold and frequency. The VCA compressorincludes an OverEasy switch for soft- and hard-knee processing, and aSlow button that changes the speed of the compressor. The 1086 alsoincludes the dbx PeakStopPlus two-stage limiter to keep the output fromoverloading the recorder input.

Drawmer MX60. Officially called the Front End One, the MX60 ($629)has all the traits of a well-equipped voice processor (see Fig. 3). Therear panel includes +4 dBu balanced and -10 dBV unbalanced 11/44-inchline inputs, a mic input, and an insert jack. An instrument-level inputis strategically placed on the front panel. The MX60 can simultaneouslyhandle balanced and unbalanced line-level inputs, and balanced andunbalanced line-level outputs. Together, these features allow you touse the MX60 for level conversion (see Fig. 4).

The dynamics section of the MX60 includes a gate,compressor/limiter, and de-esser. The design of the MX60's VCAcompressor is based on the DL241 and MX30. The three bands of EQinclude a fully parametric mid band (with frequency, bandwidth, andcut/boost controls), as well as high- and low-shelving filters. Next isTubesound, which has three separate adjustable bands of saturation: Locovers 350 Hz and below; Mid handles 350 Hz to 2 kHz; and Hi is 2 kHzon up. Each band has a Drive control that ranges from 0 to 11.

The MX60 has a fixed-threshold (+20 dBu) prefade limiter before themaster output fader. The limiter isn't user selectable-soft limitingbegins automatically at +6 dBu-so you really have to watch how youmanage the various gain stages to avoid unwanted limiting.

Finally, the MX60 has no power switch. To avoid sending damagingcurrent spikes to your mics when you turn on your power strip or lineconditioner, the unit's phantom power comes up and decays slowly.

Focusrite Platinum VoiceMaster. The Platinum series is Focusrite'sforay into the price realm of the average personal studio. Whileproducts in their Red series cost upward of $2,500, the units in thePlatinum line are priced well below $1,000 and contain manywell-thought-out features based on the more expensive models.

For example, the Platinum VoiceMaster ($749) contains a Class Adiscrete transistor mic preamp with a frequency response of 10 Hz to200 kHz (with -1 dB variance), which exceeds the human hearing range.Another notable feature is an opto compressor that includes a Treblecontrol for reintroducing high frequencies into heavily compressedsignals (see Fig. 5). The threshold and release times have variablecontrols, while speed and ratio are set with two-position buttons:Attack time is set using the Fast button, while Hard Ratio gives you achoice between high (6:1) and low (2:1) ratios. In addition, theVoiceMaster has an opto de-esser as well as a tunable saturation stage.An expander/gate and EQ section round out the feature set.

VoiceMaster's back panel includes mic- and line-level inputs, aninsert jack, and an XLR output that lets you take a signal out beforeit gets to the de-esser (see Fig. 6).

HHB Radius 40. HHB distributes TL Audio's Ivory Series 5051 ValveProcessor in North America under the name Radius 40 Tube VoiceProcessor. (See the review of the 5051 in the December 1998 issue ofEM.) Designed primarily for vocals, the Radius 40 ($749) is the onlyunit in this group without a built-in de-esser. A sidechain jack isincluded on the back panel so that you can de-ess the old-fashionedway.

Inside the Radius 40 are three tubes. The first is used for theinput stage as part of a solid-state/tube hybrid circuit. The secondand third are used in the compressor and 4-band parametric EQ stages.As the levels through the tube stages increase, the amber Drive LEDilluminates. The red Peak LED signals that there is less than 5 dB ofheadroom left.

The input section includes a front-panel 11/44-inch jack that canaccept instrument-level and line-level inputs, and a switchable 90 Hzlow-cut filter (see Fig. 7). The back panel has a mic input, 11/44-inchunbalanced and XLR line-level inputs, the sidechain insert, a link jackfor synchronizing the VCAs of two Radius 40s, an input-level switch,and XLR and 11/44-inch unbalanced outputs (see Fig. 8). Interestingly,the Radius 40 lacks a phase switch, which seems an odd omission for adevice that can be linked into a stereo configuration.

The Radius 40 has a proprietary solid-state compressor called atransconductance amplifier, with four fixed attack and release times.Threshold, ratio, and gain have continuously variable controls. Theattack ranges from 0.5 to 40 milliseconds, and the release times arefrom 40 ms to 4 seconds. The compressor can be switched out of thesignal path using a front-panel button.

Each of the EQ bands has four fixed bandwidths and a variable levelcontrol. The two mid bands have fairly wide bandwidths so thatfrequencies in adjacent bands overlap. The EQ section follows thecompressor in the signal path but can be easily switched ahead of thecompressor using a button on the front panel.

LA Audio PS-1. The most expensive voice processor of the bunch, thePS-1 ($850) has a mic preamp (with high- and low-cut filters), afull-featured compressor (including variable control over attack andrelease times), a parametric EQ with two variable mid bands, anexpander, and a de-esser (see Fig. 9). In addition, the PS-1 can befitted with an optional A/D converter.

The back panel gives you a good deal of flexibility by havingseparate inputs and outputs on the EQ and dynamics sections (see Fig.10). The input section has mic, line, and DI inputs and a 11/44-inchoutput. The dynamics processor has a line-level input and output aswell as stereo-link and sidechain jacks. The EQ has line-level inputand output, and the Output stage has a 11/44-inch input next to the XLRoutput. All this I/O flexibility means that you can shorten the path tothe recorder by using the preamp's 11/44-inch line output, use thedifferent effects independently, or reorder the effects in the signalchain.

SONIC FINDINGS The best way to test a voice processor is on a voice.I asked singer/songwriter/ guitarist Jill Garellick of Cactus Motel tobe my test subject because of her dynamic singing and strumming style.I also recruited engineer Myles Boisen for his ears and gear.Garellick's guitar and voice were recorded separately using a matchedpair of mics so that I could use two voice processors per take. Irecorded the guitar using a factory- matched pair of Oktava MC012microphones and tracked the voice with a matched pair of '70s-eraNeumann U 87s. Each mic went through a single processor and thendirectly to tape.

To further test the behavior of the units, I ran a few differentinstruments through them, including guitar, bass, theremin, keyboards,and drum machine. I also routed tracks from tapes (music and spokenword) through them to see how each device reacted to various trackingand mixing conditions.

Pump up the volume. Each of the mic preamps in the collection hasphantom power and a low-cut switch, and all but the Radius 40 havephase reversal. Other than features, what really sets these preampsapart from each other is how they sound; when heard side by side thedifferences are remarkable.

The combination tube/solid-state design of the Radius 40 mic preampgives it more coloration than all but one of the other preamps, butwith an added throatiness. Boisen described its sound as "warmer ormurkier, depending on your taste," a quality that is no doubt due tothe tube influence. The Radius 40 does, however, have a pronouncedupper midrange that helps counteract its preamp's slight muddiness.

Heard on its own, the Radius 40 preamp is noticeably fuzzy. Even inthe clean settings, the contours of the voice sounded like they werewrapped in gauze. Once the Radius 40 tracks were placed in the mix,however, they sounded great. The Radius 40 guitar track blended wellwith the vocals tracked through the other processors, and vice versa.With that in mind, I'd be less conservative with the degree ofprocessing next time I use the preamp. Although it doesn't have as much"clean" headroom as the other preamps, you can push the overall levelmuch further.

In contrast to the Radius 40, the PS-1's preamp had a smoother highend and an impressive clarity. The high harmonics of the guitar reallycame through. As far as tonal coloration, its preamp was neither themost nor the least transparent of the five units. Compared with theMX60 and the VoiceMaster, I detected a slight buzziness around thevoice, similar to that of the Radius 40 but far less prominent. And thePS-1 preamp seemed to have quite a bit more headroom.

The preamp in the dbx 1086 was a little less satisfying to use. Thestepped input gain was problematic at times: there were drastic changeswith each step, and sometimes the level I wanted was between thenotches. Adjustments to the level often affected the behavior of thecompressor, making it more difficult to predict the response. Andsimilar to the Radius 40, the 1086 had the least amount of headroom andrequired a greater input gain to get a level on tape that matched thePS-1, VoiceMaster, and MX60.

The 1086 was the least transparent preamp of the bunch. Hearing iton its own, I could detect a little coloration. In side-by-sidecomparisons, however, it sounded somewhat two-dimensional, and thecoloration was far more pronounced. But coloration is not necessarily abad thing. EM associate editor Brian Knave used the 1086 preamp tosmooth out an otherwise harsh vocal sound. One might say that the1086's preamp has a "soft focus" effect on the voice.

On the other end of the spectrum is the preamp in the MX60. Both thevoice and the acoustic guitar sounded fantastic through it. Boisennoted a slight boost to the low mids, which added a bit of lumpiness tothe sound. You have to mind the MX60's input level carefully, otherwisethe prefade limiter starts working on the sound. If you want to avoidthat stage altogether, take the +4 dBu signal out of the balanced11/44-inch insert jack on the back panel. The MX60 Class A preampincludes a phase reverse switch, a 100 Hz low-cut filter, a Brightnessbutton, and a -20 dB pad.

Of the five mic preamps, the VoiceMaster is the most transparent. Asthe other Class A preamp in the collection, it's easy to see why. Thehigh end is crisp and clear, and the frequencies are more evenlydistributed than in the other preamps. I compared the VoiceMaster tothe Focusrite Green Series Dual Mic Pre, and the sound was remarkablysimilar.

Total transparency may not always work for a session, so it's niceto have the ability to gradually color the sound to taste. Both theVoiceMaster and the MX60 give you a tunable saturation stage that, onthe voice, sounded convincing in very small doses. And while we'rediscussing saturation, coloration, and buzziness, let's see whathappens when we run line- and instrument-level instruments throughthese devices.

Instrumental connections. Because TL Audio/HHB's and Drawmer'sdesigners were nice enough to put instrument jacks on the front panels,I'll begin with the Radius 40 and the MX60. The 11/44-inch jack on thefront panel of the Radius 40 can handle just about any signal. That'sgreat, because going behind your rack each time you want to plug in aninstrument is inconvenient. Keep in mind that it's an unbalanced input:if your keyboards are across the room and you need the RFI protectionof a balanced line, you may end up going behind your rack after all. Besure to have a TRS-to-XLR cable as well because the Radius 40'sbalanced input uses an XLR jack.

Like the Radius 40, the MX60 has an unbalanced 11/44-inch input onthe face, though it is meant for instrument-level signals only. Theback panel has +4 dBu balanced and -10 dBV unbalanced 11/44-inch inputjacks, and you can use both of them, as well as the balanced andunbalanced outputs, at the same time.

In its back-panel input section, the PS-1 has jacks for line- andinstrument-level signals, or you can go directly into the dynamicsprocessor, equalizer, and output section if you like. The 1086 and theVoiceMaster do not have a built in DI. The only line-level input on the1086 goes directly into the dynamics processor.

Dynamic differences. There are bound to be compromises when you jama multitude of effects into one box priced under $850. This is mostnoticeable in the feature sets of the dynamics processors. As far asgetting a full-featured compressor, the PS-1 and Radius 40 come theclosest. They both offer control over attack and release times, as wellas ratio, threshold, and make-up gain.

Besides having variable threshold (-30 to +20 dB) and ratio (1:1 to20:1), the PS-1 is the only one of the bunch with continuously variableattack and release controls. In addition, it has a Gain knob with 20 dBcut or boost, as well as a hard/soft-knee button and a bypass switch.The PS-1 compressor can be used independently of the other stages, andit has a link function for stereo use.

However, the PS-1 compressor is more challenging to use than theother compressors in this group. The controls are very touchy, and theslightest movement noticeably affects the sound. And to get the meterbar to match what I was hearing, I had to push the input gain more thanwith the other units.

On the other hand, the dbx meters tell you the story right away.Although the 1086 has no release-time control, it is smoother soundingand more musical than the PS-1. You adjust attack and release timesautomatically on the 1086. By engaging the Slow button, you can extendthe attack and release times for instrumental applications. But havingso little control over the speed made the 1086 compressor challengingto use; getting the right setting was sometimes difficult due to thestepped controls.

Setting up the VoiceMaster's opto compressor, by contrast, was quickand easy. It has only two ratio settings to choose from-Soft Ratio(2:1) is intended for vocals and Hard Ratio (6:1) for instruments.However, the soft setting was perfect for acoustic guitar as well asvoice. And having a variable release time helped in getting the rightsound for electric guitar and bass tracks. The Treble feature proveduseful in reinstating high frequencies postcompression.

The VCA compressor in the MX60 was a tad smoother than theVoiceMaster's opto compressor. Although the attack and release timesare automatically controlled, I had no trouble getting a natural andmusical sound. Besides a variable threshold control, the MX60 has acontinuously variable ratio. The MX60's compressor cannot be linked forstereo use.

The Radius 40 has one of the smoothest, most gentle compressors ofthe group. It was easy to dial in a setting on both the guitar andvoice, using medium-slow speeds and a ratio of about 8:1. Drum samplessounded especially beefy through this compressor. And having the optionof switching the EQ in front of the compressor helped even out thecompressor's response to bass-heavy signals.

Mind expansion. All but one of the voice processors in this groupplace the expander/gate before the compressor. The Radius 40'sexpander/gate is at the very end of the signal chain and is by far thesimplest of the bunch. It is controlled by a single knob with a rangeof off/-50 to -20 dB. This expander/ gate acts as a gate in the highersettings (fully clockwise, between -30 and -20 dB). However, thisresults in envelopes getting sped up and sharp transients beingchopped-so much so that, when set at -25, it gave drum samples areverse-sounding attack.

Although the Radius 40's expander/ gate control comes after themaster output control in the signal chain, the knob is located next tothe make-up gain in the middle of the front panel. This clued me inabout how to use it. When I really pushed the input and make-up gainstages, I could use the expander/gate to keep the noise of thetwice-boosted signal from coming through. Still, I missed havingseparate control over the speed of the gate. Other than this oneapplication, I had a difficult time using this particular expander/gatesuccessfully.

Compared with the Radius 40, the 1086's expander/gate is not onlyeasy to set up but sounds fantastic. Threshold and ratio are your onlycontrols, and with the 1086, that's enough to shape percussion and keepnoise to a minimum. The threshold range is off/-80 to +15 dB, and theratios span 1:2.1 to 1:8 (although they are mistakenly printed as 1.2:1and 8:1 on the front panel). At times, the gate seemed a little touchy,probably due to its automated program sensitivity.

LA Audio refers to the PS-1's expander as Noise Reduction. This mayseem slightly confusing at first, but as a downward expander, itdelivers on its promise. The Noise Reduction function has a presetratio equivalent to 3:1, a threshold control that ranges from -70 to 0dB, and a two-position, switchable release time (fast or slow). Theattack is preset with a soft-knee envelope that sounded natural oneverything I ran through it. An amber light lets you know when thesignal is above threshold. The back-panel link connector allows you tocoordinate the expanders of two PS-1s for stereo operation.

The PS-1's knobs are highly sensitive, requiring very subtlemovements to get the right timing. However, I had no trouble dialing ina natural decay each time. As with the PS-11's de-esser and compressor,you can use the expander independently of the EQ and input section. Thethree dynamics processors share the same I/O, so you can't plug intoeach of them independently.

The gate Drawmer put in the MX60 is actually more of a downwardexpander. The MX60's gate has two speeds (fast and slow) and a variablethreshold (off to +20 dB). As with the other gates, changes inthreshold will vary the speed: when you dial toward a higher thresholdthe gate moves quickly, and lower threshold settings (the MX60 descendsto -70 dB) slow the gate down.

It was often difficult to make the MX60's gate sound transparent.However, when I did find the right setting for an application, itworked well. The combination of a low threshold with the slow speedsetting was perfect for fattening a noisy drum machine. In thisparticular instance, the threshold control made it fast and easy tofind a decay that sounded musical.

The VoiceMaster's Noise Reducing Expander includes a switch tochoose either gating or expansion. The variable controls includeThreshold (-40 to +10 dB) and Depth (0 to Full). A 4-LED bar graphindicates the amount of reduction being applied to the signal. TheVoiceMaster's expander worked well with vocal tracks (both spoken andsung). But when the gate was engaged, the speed was so fast that it wasdifficult to find a good setting using only threshold and depthcontrols. This particular gate could use a speed control.

EQ review. Like dynamics processing, a parametric equalizer is handyfor both tracking and mixing. The most useful parameters in themidrange frequencies are adjustable bandwidth, sweepable frequencyrange, and cut/boost control.

The PS-1 possesses the most dramatic-sounding EQ of the bunch, withtwo full-featured mid bands in addition to high and low shelving. Eachof the PS-1's mid bands gives you control over a range spanning 60 Hzto 20 kHz. The MX60 has a similarly wide frequency range of 150 Hz to16 kHz. However, this range is packed into a single parametric bandwith shelving filters on either side.

The EQ in both of these units worked so well that I wanted to usethem on everything! And it was easy to zero in on the frequency Ineeded. These EQs are so accurate that you don't need much boosting orcutting to hear a change in the sound.

The EQ on the Radius 40 was far less dramatic, but no less musicalthan the PS-1 or MX60. The combination of wide bandwidths andoverlapping frequencies made this EQ particularly useful. Having a tubestage in the EQ is also a plus if you want to add extra coloration.

Although the 1086 has no EQ section, the preamp has avariable-frequency low-cut filter. I didn't find the dual-band Detailfeature particularly useful on voice or acoustic guitar, though it mayprove useful with other miked instruments. I was disappointed that the1086 had no line-level input before the dynamics processor: I suspectthat Detail might work well on thin-sounding electronic instruments,but there's no way to try it.

The voice-optimized EQ on the VoiceMaster has five controls,cryptically named Warmth, Tuning, Presence, Absence, and Breath. Thissection worked well once I figured out how each of the controls works.Breath is a 10 kHz shelving filter for cutting or boosting the "air"frequencies. Tuning and Warmth, combined, are like a low-frequencyparametric EQ. Tuning determines the center frequency (between 120 and600 Hz) that Warmth cuts or boosts. Presence is a fixed bandwidthcut/boost control with a peak at 1.5 kHz. The Absence button has acenter frequency of 4.5 kHz and is intended to further attenuate themidrange by 6 dB. The area around 4.5 kHz is where the voice can soundharsh.

On the units that have equalizers, the EQ is placed after thecompressor in the signal chain. Sometimes, however, you may want theequalizer before the compressor. For example, if a specific frequencyis causing the compressor to pump, you can use the EQ to cut theoffending frequency. Two of our voice processors give you this power:the Radius 40 has a front-panel button that lets you move the EQ aheadof the compressor, and the PS-1 allows you to repatch the order on theback panel.

Mind your s's and z's. The de-esser comes after the compressor onthe 1086 and the VoiceMaster, and before the compressor on the MX60 andthe PS-1. In fact, on the PS-1, the de-esser is the first stage afterthe input. On the MX60, the de-esser comes between the gate and thecompressor.

The advantage of having the de-esser before the compressor is thatyou can remove sibilant peaks that cause unwanted compression. On theother hand, having the de-esser last in the chain is useful forremoving increased sibilant frequencies caused by extreme compression.Unfortunately, none of the units in the group allow you to switch theorder of the de-esser and compressor in the signal chain.

The de-esser on the 1086 uses a variable-frequency highpass filterto attenuate the sibilant frequencies. Consequently, it acted the mostlike a traditional sidechain de-esser. At the extreme settings I couldhear the ducking action on sharp transients, but in normal usage thisde-esser was more useful than those in the other processors. The 1086de-esser followed the dynamic contour of the voice nicely, and at timesit seemed to reach out and grab the sibilants. The 1086 also has a bargraph that visually indicates (from -1 to -30 dB) the amount ofde-essing being applied to the signal.

The 1086 has a threshold and frequency control for its de-esser. Thethreshold is numbered from 1 to 10, and the frequency ranges from 800Hz to 10 kHz. Having the option of going below the normal sibilantfrequencies is useful for instrumental applications.

The PS-1 de-esser seemed less drastic but worked well, perhapsbecause of the sharp cutoff of the filter. The PS-1's de-esser uses alowpass filter, sweepable from 800 Hz to 8 kHz. The Listen functiongave this unit an advantage over the others. When Listen is active, youhear only what's above the filter, making it easier to locate the exactsibilant frequency. Boisen suggested that the precision of the PS-1de-esser would make it harder to make a mistake with this effect.

The VoiceMaster's opto de-esser was the most subtle of the group. Infact, it's not a true de-esser at all, but rather a lowpass filter witha frequency range from 2.2 kHz to 9.2 kHz. One interesting feature ofthe VoiceMaster is that it includes an aux output that lets you tap thesignal before it reaches the de-esser. That way you can use a de-essedsignal for effects (such as a reverb), while sending the non-de-essedsignal to tape. Because the VoiceMaster's de-esser is last in thesignal chain, both signals will have identical amounts of dynamic andEQ effects.

The MX60 had the most easily audible de-essing effect of the group.Although Drawmer says the work is done by "intelligent circuitry," itsounds more like a shelving filter than a true de-esser, and it didn'ttake much attenuation to begin coloring the sound. Instead of having avariable frequency control, you choose between Male and Femaletonalities. I found that the MX60's style of de-essing worked better onspoken- word material rather than on singing.

Tube or not tube. The MX60 and the VoiceMaster have features thatemulate the "warm" sound of tape saturation and tubes. On theVoiceMaster, the saturation stage is before the compressor, whereas onthe MX60 it's after the compressor and the EQ. A little goes a long waywith each unit.

The VoiceMaster's Vocal Saturator creates smooth-sounding distortionat lower gain levels. Crank up the saturation level, and you get adelicious overdrive without a lot of noise or fuzz. The VoiceMastergives you the option of tuning the saturation. With the Full Bandwidthbutton in, you get a warm, wide-band distortion. With Full Bandwidthout, you can use the Tuning control to precisely select the upperfrequencies (between 1.4 and 7.2 kHz) you want emphasized. The Driveknob ranges from Clean to Unclean. I had to keep the Vocal Saturatorbutton engaged, even when I didn't intend to use the effect. Wheneverything but the Saturator was in the signal path, the VoiceMaster'sself-noise became more noticeable. Once I added the Saturator into thesignal path (with the level set to Clean), the hum went away.

The MX60 can be pushed much further than the VoiceMaster, and withthe help of the 3-band Tubesound, I was able to tune in an amazinglayer of distortion. It also helps that there is an extra gain controlbefore the Tubesound stage. Having three frequency bands to work withmade it much easier to get results that sounded convincing on voice.But then again, Tubesound sounded great on everything I put throughit.

The most over-the-top sound came from the Radius 40 and its threetubes. With the input and compressor gains nearly maxed, and the EQsent precompressor, I could fine-tune four frequency bands ofdistortion, which was enough to make the thinnest keyboard soundgigantic. You won't use this effect every day, but it's nice to knowyou have it.

Direct to digital. What could be handier than having digitalconverters built into your processor? The dbx 1086 and the LA AudioPS-1 both have optional A/D converters, though neither of the units Ireviewed had the converters installed.

The dbx 1086 includes a space on the rear panel for the new 504XDigital Output card ($400). This card uses the proprietary dbx Type IVA/D converters for 16-, 20-, and 24-bit resolution. The 504X includesboth AES/EBU and S/PDIF connectors, with a switch to select the format.Additional buttons allow you to select 16- or 20-bit word lengths and44.1 or 48 kHz sample rates. The 504X also has word-clock input andoutput, so you can slave the unit or use it as the master clock.

The 1086's front panel has a three-position Dither switch for theconverter card. In the Off position, the unit sends a 24-bit signalthrough the digital output. The two types of dither are TPDR and SNR2.There is also a Shape button to select Type 1 or Type 2 noiseshaping.

The optional A/D converter for the PS-1, the PS-DR ($329.95), isalso 24-bit capable. It has word-clock input only, but it adds aToslink optical output to the S/PDIF and AES/EBU connectors. Like the504X, the PS-1 can run 44.1 and 48 kHz sampling rates, but you candither only to 16 bits. The PS-1's digital card includes an additional11/44-inch audio input, so you can run two processors through onedigital card. Keep in mind that it must be installed at the factory;you may be better off buying the PS-1D ($1,149), which has the cardpreinstalled.

These two units aren't the only voice processors in this price rangethat have digital capabilities. The MindPrint En-Voice uses the DI-Mod24/48 card ($249), which has digital input and output capabilities viaS/PDIF connectors. This means that you can use the DI-Mod to run atrack from a digital recorder through the En-Voice's tube compressorand back to the digital recorder. The DI-Mod also includes an extra11/44-inch input so that you can use both digital channels of theS/PDIF connector.

BALANCING ACT As with any piece of gear you use, a voice processormust match your style of working. Deciding on which features you needmost will help you pick the processor that's right for you. If youprefer quick-and-easy results over flexibility, you'll want a unit withone or two buttons per effect. If you like to tweak, however, thenfeatures and knobs may beckon you.

HHB's Radius 40 is the easiest to use of the five voice processors.The compressor and EQ set up quickly, and the tubes give them a nicesound. The tube/solid-state hybrid mic preamp gives the voice a bit ofan edge that helps it sit well in the mix. Although I didn't find theRadius 40's expander/gate that useful, and it didn't have a de-esser, Ilove that you can easily put the EQ before the compressor. It may becalled a tube voice processor, but it works wonders on instruments, aswell.

The dbx 1086 has several big things going for it: you can use themic preamp and dynamics processor separately; the expander/gate andde-esser are fantastic; and the optional 504X A/D converter allows youto go direct to digital. The mic preamp had the most coloration of thegroup, but the competition in this area was stiff. The dynamicsprocessor is easily the high point of the 1086.

The LA Audio PS-1 has the most open architecture of the fiveprocessors. The controls for each effect are very sensitive, so it maytake you a little longer to get the sound that you're after. But it'sworth the trouble. The EQ section is top-notch, the mic preamp soundsgood, and the compressor is quite powerful once you get a handle on it.The front and back panels are easy to navigate, and I appreciate havingthe flexibility of using the dynamics processor, EQ, and mic preampindependently. And you can add an A/D converter to the PS-1 fordirect-to-digital recording.

Drawmer's MX60 straddles the line between ease of use andtweakability. The compressor is very smooth and sets up easily; thegate is a little touchy, but it's musical once it is locked in. Myfavorite feature on the MX60, however, is the 3-band Tubesound, whichmakes this processor stand out from the pack.

The Focusrite Platinum VoiceMaster has the clearest-sounding micpreamp (it's Class A) and one of the smoothest compressors of thebunch. Each effect dials in quickly and performs well. The combinationof a transparent preamp and a saturation stage gives the VoiceMaster awide tonal palette that works well on just about anything you want torun through it.

NOT JUST FOR VOICE ANYMORE By using a single-channel voice processorrather than a mixer channel, you get the shortest distance from sourceto recorder and the best signal-to-noise ratio possible. And because avoice processor can be just as useful for mixing as for recording, it'sa worthwhile investment.

Gino Robair is an associate editor at EM. Special thanks to MylesBoisen, Jill Garellick, Jeff Casey, Steve O., Brian Knave, and LauraForlin, and to Elliot Garellick for his patience.