This review originally appeared in Tape Op Magazine.

Whether to work with a control surface or not is a personal decision. If you do prefer to have faders, knobs and buttons under hand, the one thing we will all agree on is that the integration between the surface and the software has got to be right.  Tight integration is why we still have “dedicated” control surfaces designed for specific software DAWs on the market. Euphonix, however, has developed their Artist Series control surfaces to work simultaneously with a slew of applications, even bridging between audio and visual worlds. These applications include Apogee’s Maestro for Duet, Ensemble and Symphony 64, Apple’s Final Cut Pro, Logic Pro and Soundtrack Pro, Metric Halo’s MIO Console, MOTU’s Digital Performer, Steinberg’s Cubase and Nuendo, Avid’s Pro Tools HD, LE and M-Powered, Abelton’s Live and Propellorhead’s Reason. When you think about it, this cross-platform integration really shouldn’t work, yet at the push of a button you’re working on the same surface in another program with nearly flawless integration.  For this alone Euphonix’s multiple design awards are well deserved.

The Artist Series consists of the MC Mix, the MC Control and the MC Transport, which can be used together in any combination or alone. They hitch together beautifully, creating a fully modular product line. These surfaces are so versatile that a single user’s review really can’t cover it all, and there is no way I could (or would!) ever work in enough applications and situations to fully explore the wide range of ways one might use the MC series.

With that said, I will be focusing mostly on my use of the surfaces (and particularly the MC Mix) to do stereo ixing in Pro Tools – a scenario in which I imagine many Tape Op readers also regularly find themselves.  Unfortunately, Pro Tools is the one DAW with which Euphonix has had some challenges implementing integration.  Why?  Because – and understandably so – Avid sells their own dedicated control surfaces, and the Euphonix software system, EuCon, requires that the “third party” open up their software codes for integration (aside from Live and Reason, the rest of the software developers listed above have opened their codes for integration with EuCon).  For Pro Tools integration, however, Euphonix must use the HUI protocols that Digidesign originally developed along with Mackie, and the HUI protocols have built-in limitations that make using multiple MC surfaces a little tricky.  In a nutshell, the issue is that when using more than one eight-channel MC Mix with Pro Tools, the MC Mix wont be able to splay plug-in functions out across multiple surfaces. This means that you’re limited to using one surface (typically the left-most unit) for calling up and manipulating plug-in parameters. Similarly, banking faders can be confusing when using an MC Mix and an MC Control (total of 12 faders). So, for now sticking with eight channels may be easiest for Pro Tools users, and we can only hope for better integration down the road so that it functions as seamlessly as it does with all the other programs.

Yet, even with these Pro Tools-specific limitations, I still chose the MC Mix over any other control surface in its price range. Here are the main reasons: The faders are exceptionally smooth and don’t clack and jiggle; they are the same high-resolution faders you’ll find throughout the Euphonix line.  It’s really pro stuff, and writing automation with them feels both fluid and solid like a good fader should.  The unit’s footprint is sleek, comparatively small and especially designed to work well laying on a console or between a keyboard and a monitor.  In fact, if you have Apple’s new aluminium keyboard, it’s as if they form one unit functionally and aesthetically.  Ergonomically, this integration between keyboard and surface is so much easier on my shoulders and neck over the course of a long mixing day.  The LCD screens are easy to read and don’t blind you. Once you get used to them, there’s a lot more info being fed back than you may first realize.  Another reason I chose the Euphonix surface is that if I decide to move to another DAW, the MC Mix is already integrated; I get the feeling this technology is going to be developed, not outmoded by a new model.  And, most importantly for me, this is the only control surface under $10,000.00 that allows you to select automation modes in Pro Tools without touching the mouse.

That last point is at the crux of the MC Mix and MC Control’s integration with Pro Tools because reaching for the mouse to get in and out of automation modes breaks the spell of the tactile connection to the mix.  More simply, I can keep my elbows on the desk, my head in the mix and the mouse “over there.”  Automation mode selection is especially important if you’re using the surface on top of a larger console. If you think about it, the ability to select automation modes on the Pro Tools-dedicated Icon consoles (and the Pro Control before it) is what really bridges the gap between the Pro Tools and, say, the SSL tactile experiences.  Now, for as little as a grand you can have that experience…well, almost.  The one difference is that the MC Mix doesn’t accurately display exactly which of the many Pro Tools automation modes you’re in, so you do have to look up, or over, at the screen.  It’s certainly not a deal breaker, but worth mentioning in the hope that this visual feedback will come along one day soon on these surfaces.

Working/playing on the MC Mix is intuitive, fast and fun.  I was amazed at how quickly I was moving around on it, and after about an hour I was not willing to work without it again.  The faders are easy enough to understand, yet the touch-sensitive rotary encorders at the top of the unit are where you’ll discover your inner octopus.  With just a few buttons along the left side of the unit you can shift the work these encoders do between pan, aux level, eq controls, dynamics controls, insert selection and more.  The LCDs follow the change in mode with clear visual feedback that helps you know just what you’re doing.  There is also a “channel mode” which allows you to quickly manipulate individual plug-in parameters on these knobs, and I was surprised to find how much more willing I was to automate plug-in parameters once I had tactile control.  Want to lighten up on some compression during a bombastic moment? – select your channel, enter channel mode, pick the plugin by pressing the corresponding encoder (they’re also push buttons) and there you go.  I was making this series of moves in under three seconds after just a few tries.  Want to add a little volume ride across that section, too? – select your preferred automation mode and move the fader.  You can do all of this without touching the mouse.

There are a ton of other standard features on the MC Mix that I’m assuming you’ll all expect to be there, such as solo and mute, record arm and channel select buttons.  There are also ‘select’ and ‘on’ buttons associated with each rotary encoder.  Once you get a handle on what these do, they offer up deeper levels of software control that become intuitive with use.  Like any control surface, there are also buttons for moving the faders across a session in banks or individually, but Euphonix has also built in the ability to freeze a track to a particular fader and map non-sequencial tracks to the faders.  If you’re feeling fancy, you can use time-line markers to call up different maps – something I could see film mixers using to help manage enormous track counts and elaborate mixes.

Even though I found a single MC Mix (just eight faders) to work best with Pro Tools, when I did hook up three of them together the effect was that I was suddenly sitting at a twenty-four channel console, not a string of control surfaces.  For those of you using applications that can take advantage of the EuCon software integration (see the www.euphonix.com for updated info), the temptation to link many of them together will be strong.  It’s worth noting, too, that buyng two MC Mix units gives you a sixteen-channel control surface for $2000.00, a size and price-point that’s (surprisingly) absent in the market otherwise.

The MC Control and MC Transport open up all kinds of possibilities for further configuring an interface and work-flow to suit your individual needs.  The Control is a fascinating unit that features four faders, eight assignable rotary controllers, a jog wheel and transport control, twelve programmable softkeys and a touchscreen with extremely deep programmability.  The MC Control is less about tactile control of mix functions and more about building customizable command chains (macros) that can massively speed up your workflow.  While the Control didn’t meet my needs as a mixer as much as the MC Mix (they’re well named), I was fascinated to find myself building macros that would guide me through the menus and submenus of Pro Tools with the simple tap of a button.  Those of you who do deep level editing and programming in your DAW are going to want to check out the Control right away.  It’s a deep and powerful interface that’s begging for you to design a custom workflow and leave the mouse “over there.”

The MC Transport has an enormous and very positive feeling jog wheel and outer shuttle ring, full transport and navigation controls, six assignable soft keys, timecode display and a numeric keypad.  Adding the MC Transport to either the MC Control, the MC Mix or both will allow you to nearly eradicate the mouse from your workspace.  While the ability to move through your sessions with ease is an obvious advantage, the soft keys allow you to change the functionality of the jog wheel and shuttle ring on the fly.  It’s a deceptively powerful control surface all on its own, and one that will keep many mundane editing tasks from being so thoroughly annoying.  The more you play with the MC Transport, the more you realize its flexibility.  There’s very little you can’t do with it.

What I love about Euphonix Artist Series is that it lets you decide which controls you want, how many of them and how much you want to spend.  It a truly customizable modular system, and they’ve now included the MC Color, which is designed for film colorists.  Their website has excellent information and videos to help you understand which surfaces would make the most sense for you, and if you’re considering any of these, I encourage you to spend some time watching the videos that are specific to your applications(s).  You’ll learn a lot. For those of you who use Pro Tools, do not let the current limitations of integration scare you away.  You’ll love these surfaces for their exceptionally pro feel, look, flexibility and expandability.

This article originally appeared in Tape Op Magazine.

Brent Averill Enterprises (BAE) is perhaps the best-known builder of Neve-style preamps, and for good reason as their designs indeed capture the behavior, tone and appearance of Neve preamps beautifully.  For years I have used a rack-mounted pair of their 1073s, and they have performed flawlessly, providing that fat and warm, yet open and dynamic sound that has defined the vibe of countless records we hear every day. For those who aren’t familiar, a Neve 1073 module has a mic preamp, a line-amp (with it’s own dedicated transformer) and an eq with a high shelf fixed at 12k, a variable mid bell-curve section with fixed q, a variable low-shelf section and a variable high-pass filter.  They are classic modular preamps, first released in 1970 as part of the A88 mixing console, and the originals have gone on to become legendary, collectible and very expensive. To take on the task of recreating the sound of Neve’s classic 1073 preamps is always a bit of a tight-rope walk, but to try to expand on the design while maintaining the original vibe and sound is to walk without a net.  That’s what BAE has done with their new 1023 preamp/eq.

Like the original 1073, the 1023 is completely handwired using Carnhill (St. Ives) transformers. It has the exact same mic/line preamp as the 1073, but with significantly more frequencies in the mid and hi sections.  Aside from simply offering more frequency settings to play with, these expanded eq sections also allow you to play the mid bell curve directly against the high and low shelves where they overlap.  This capability opens the tone shaping possibilities in very interesting and musical ways.

On the middle section of the 1023 eq you’ll find two additional lower frequency settings and three higher ones than you would on a 1073, which gives the 1023 the following mid eq points: 160Hz, 270, 360, 510, 700, 1K6, 3K2, 4K8, 7K2, 8K2, 10K.  The two new low-mid points (160Hz and 270Hz) overlap with the low-shelf’s frequencies, allowing for some fun tone carving in the warm region.  For example, while boosting 160Hz with the mid band and cutting 220Hz on the low shelf you can achieve a very tight rise in the lows that doesn’t overpower in the deeper frequencies or get too muddy up around 300Hz.  The sound is quite different than simply boosting 160Hz.  This particular eq setting is really fun for fattening up distorted electric guitars, warming up female vocals, or getting a floor tom to growl in a new way.  The low end is always tight, punchy and satisfying.

On the high shelf, rather than the fixed 12kHz shelf of the 1073, you’ll find settings at 10K, 12K, 16K, 20K and 24K.  The expanded high frequencies in the mid-section start to make sense when you realize that you can really play the mid section against the high shelf, just as you can with the low.  With drum overheads, for example, try boosting 10kHz on the mid section while cutting 20kHz on the high shelf and you’ll get an increased sizzle with a decrease in the air region that is reminiscent of some vintage recordings.  Or, do the opposite to control brash cymbals while adding some excitement in the air region.  The possibilities are pretty limitless, and experimentation is fun and rewarding.  As to be expected, the highs are smooth and musical, just as a Neve should be, but the added frequencies on the high-shelf make the 1023 more versatile and fun to use.

The 24kHz setting is my favorite feature on the 1023.  I want to deviate for a moment and discuss what it means to be working with a frequency that is, presumably, outside the audible range.  First, the curve of a 24kHz shelf is going to reach down into the audible range, especially on a wide-q equalizer like a Neve.  As you turn it up or down, it will drag lower frequencies along with it.  Second, inaudible frequencies will impact the character of audible ones by way of the harmonic relationship.  What this means is that, while you might not hear what’s happening at 24kHz in and of itself, you will easily hear the impact of 24kHz on the sound of your recordings. (To further deviate, it is interesting to consider that Sear Sound in NYC has a custom console with 30kHz shelves on every channel.  Interestingly, Walter Sear stresses that the digital formats render frequencies in that region as noise, thus negatively altering the harmonic relationships. Analog tape, he argues, preserves those relationships accurately and therefore sounds better.)  The practical reason Walter Sear or BAE would put such high bands on their eqs is that the impact on the recorded music is so satisfying to the human ear.  A tiny boost of 24kHz on a female vocal brings out an ethereal quality; on acoustic guitar it helps rhythm parts occupy the realm of ride cymbals with less competition; on overheads it seems to lift a veil you may not have known was there; and on the whole mix 24kHz can bring a lot of energy and openness without harshness.  Because the circuitry is characteristically smooth in handling high frequency boosts, playing with the 24kHz shelf on the 1023 is always satisfying, even when boosting at extreme levels.

With the eq disengaged, the 1023 is indistinguishable from the BAE 1073s I’m so used to.  If you know what 1073s sound like, then you’ll know what the 1023 mic preamps and line-amps sound like.  (If you aren’t familiar with the Neve sound, expect to fall in love with the warm yet open and detailed sound.)  These are first-rate preamps; they sound amazing and handled everything I ran through them beautifully.

One of my favorite applications of the 1023 – and why I see an investment of this magnitude to be well worth it – is on an analog 2-buss chain while mixing in the box.  Running mixes through the 1023 at unity without eq can add depth, punch and width to a mix that can give you a great deal of the sonic characteristic of mixing through an analog console.  Switch in the eqs and open the top with a slight 24kHz boost, and things get really nice really quickly.  Again, when you consider this application, it’s easy to understand how the pair of 1023s I’ve had on hand have been in constant use since I got them, whether I’m tracking or mixing.

Even though the 1023’s eq is neither phase linear nor surgically exact, I’d highly recommend that mastering engineers who are looking for a “color box” check out a pair of the 1023s, as I loved their impact on full mixes with and without the eq engaged.  Mastering engineers will appreciate the added frequency settings in the mid and high eq sections, and that 24kHz setting might just be the fairy dust you’re looking for in many cases.  Combine that with the analog body and punch you get from the line-amp transformers, and it’s clear that the 1023 can bring a lot to a mastering situation where the client is looking to you to warm up and enliven mixes with an iconic analog flavor.

The 1023s come as either 10-series modules ($2975 per channel street) or as a 1U rack-mount unit ($3200 per channel street including the power supply that will power two units).  If I owned a 10-series console, I’d be looking to get at least two channels of the 1023 in there, as I know I’d reach for them all the time during tracking and mixing. If you can make the financial leap to get a pair in either format, I know you’ll find yourself using them constantly. I happily welcome the 1023 to the 10-series family.  Allen Farmelo www.farmelo.com

This review originally appeared in Tape Op.

The Pro-Ac Studio 100s are passive two-way monitors sporting a 6 ½” cone and a 1” soft fabric dome tweeter housed in a traditional looking ported cabinet. The crossover allows for typical single input use, bi-wiring and bi-amping. All wiring and components are very high-grade stuff, including oxygen-free copper cabling throughout. Since 1990, the Studio 100 has been a big seller in the audiophile market, and this model caught on as a studio monitor sometime in the middle of that decade. They aren’t exactly ubiquitous, but they’ve earned a place in the pantheon of classic studio monitors and continue to weather the ever-changing fads of speaker-design.

The Studio 100s present a smooth, full-range sonic image that emphasizes deep bass and airy highs, and they have the overall sonic character of a very transparent yet flattering hi-fi speaker. However, don’t fall for the idea that “flattering” means inaccurate or colored. To the contrary, this is a speaker that you will enjoy listening to all day every day because it is offering up a full-range, uncolored experience without any notable harshness or ringing. I have used them extensively in conjunction with Yamaha NS-10Ms, and I’d say that they are almost exact opposites of each other sonically. The NS-10s are all about the midrange and can be very harsh while the Pro-Acs are all about the deep lows and airy highs and are as smooth as can be. Switching between these two speakers is like looking at negative images of each other. Each delivers its own kind of information, and both are very helpful during a mix.

Using the Studio 100s at very quiet levels is an excellent way to get a rendering of the complete sonic image. Employing very high shelf eqs (say at 20khz on a vocal or drum overheads) at low volumes is very revealing in a way that the NS-10s never seem to deliver. Similarly, the woofers are exceptionally responsive to deep bass at lower volumes (Tony Levin’s low B string was well represented, for example). Taking this quiet approach with the Studio 100s had me making decisions about the extended frequencies I’d often have left for mastering.

Turn them up and you’ll get way more low-end than you’d expect from such a small box with a 6 ½” driver. Part of this low end must be due to the fact that the woofer can really travel. This is an idiosyncrasy that turns out to be a huge advantage, as you can use the woofer as a visual gauge of your low-end. If that cone starts to travel indiscriminately far, start fishing for the build-up, give it a fix and you’ll notice improved clarity not just in the low-end but in the whole mix. Those who use these speakers seem to know this about them: get the cones to behave and your low-end wont over-tax real-world consumer speakers.

Overall, working on the Studio 100s is a full-range experience with no noticeable dips or peaks across the spectrum. One advantage with passive monitors is that they tend to have fewer issues around the crossover point than their powered cousins, and the Studio 100s are a great example of this phenomenon. They’re very flat. Of course, pairing passive monitors with amps adds to both the complexity and freedom of developing your monitoring situation. As Pro-Ac elegantly puts it: “The full potential of these thoroughbred designs will only be realized through the use of the highest quality partnering equipment.” Translation: “Use a great amp for best results.”

If you’re still searching for your dream monitors, and the current self-powered offerings aren’t floating your boat (and many complain of the crossover issues with many of today’s monitors), I’d highly recommend looking into these speakers as a different approach from that of the current trends. They aren’t going to blow clients against the rear wall, but if you want a very refined, non-fatiguing listening experience with almost no sonic anomalies, the Studio 100s are up there with the best of the passive monitors. They’ve remained popular for nearly twenty years for very good reasons. ($2000 MSRP www.proac-loudspeakers.com) Allen Farmelo www.farmelo.com

This review originally appeared in Tape Op Magazine.

The Arturia Analog Factory Experience is the first “hybrid synthesizer” – a hardware controller and soft-synth designed together, with the controller as an exact physical representation of what you see on the screen, with all controls mapped accordingly.

The controller is, simply, the best small USB keyboard MIDI controller I’ve come across. With cream paint over a metal body and wood end-caps, it has the look, feel and weight of a high-end synth. Upon touching the keys, wheels, knobs and buttons, I knew this was an instrument on which I could really make music. The feel is smooth, solid, positive and quiet – nothing like the flimsy, clacky plastic affairs that make up the bulk of the small keyboard MIDI controller market. I believe that any instrument should, though it’s tactile interface, inspire the user on a physical level, and the Arturia certainly does. You might think of the Arturia as the Les Paul of small MIDI controllers.

Rather than offer full-on replicas of analog synths as they do with their other products, Arturia has packaged together 3500 presets derived from their digital replicas, incuding the Mini Moog V, Moog Modular V, Jupiter 8V, Phrophet V, Prophet VS, Arp 2600 V and Yamaha CS-80 V. To select the presets you use the on-screen browser, a very flexible database manager that lets you easily filter through the massive list of presets (and you can use the controller to select presets, too, abandoning the mouse all together). For example, click on Mini Moog, then on Bass, then on Aggressive, and you’ve narrowed your search down to a handful of presets. Click on Jupiter 8, Strings and Ambient and you’ve got a whole different subset of the presets to chose from. You can also skip any of the search criteria, or compile them. Perhaps you want to search all synths for sequences, or both Moogs for complex ambient pads. All presets can be tweaked and saved as a User Preset, which you can name and also assign the various search criteria. Within minutes I had modified a Moog bass and named it “Allen’s Minimoog Thumper.” It couldn’t be easier.

For tweaking, every preset has controls for volume, cutoff, resonance, LFO rate, LFO amount, chorus, delay, ADSR (on faders) as well as four “key parameters.” The key parameters control the important aspects of the preset as chosen by the programmer, allowing you to tweak the “key” aspects of the sound. I’ve found that these controls are more than enough to highly modify the sounds and create my own distinctive presets. To know what the four key parameters are for each preset, version 2.2 of the software includes four small panels on the screen describing them (e.g., Envelope Amount, Osc 1 Level, etc.). For people who want to call up some great sounds, tweak them, and start making music, the Analog Factory offers just enough control over the presets to customize them without overwhelming the user with programmability.

The sounds are amazing. Huge Minimoog basses, brash Prophet brasses, elegant Moog Modular sequences, lush Jupiter 8 string patches, bizarre sound effects, cool electro percussion stabs, angelic CS-80 organ sounds, nasal square-tooth and buzzing saw-tooth leads. warbly Arp 2600 sweeps, lacey ring-mod chimes. I’ve created whole productions of songs using only the Analog Factory (drums included) and am thrilled with the results. These are fat, thick, rich, three-dimensional tones that inspire me every time.

Yes, it’s true that there is an appreciable difference between the real-deal and these models, but I’m willing to say that Arturia has made that difference as small as I’ve seen it to date. For $300, I’m creating fat, warm, lush synth tracks that would have cost many thousands of dollars to create from analog synths. Hitting some transformers on hardware inserts during mix down adds just enough extra sonic girth to bring these tracks into full bloom. I’ll also mention that the digital system you’re using makes a big difference. On Pro Tools HD, clocked to a Cranesong HEDD, all running on balanced power and coming out of my Focal Solo6 monitors, the sound is “there” in a way that isn’t going to happen on a USB-powered mBox with headphones.

As Thom Monahan mentioned in his review of the Arturia Jupiter 8V in the last issue (Tape Op #67), the software is fairly processor intensive, especially in Pro Tools. I did find myself increasing the hardware buffer size when using multiple instance of the synth, and there were times when the controller wasn’t talking to the software. But, by the time this goes to print, Arturia may have already offered a bug-fix, or I might have upgraded to Pro Tools 8 – software is a moving target. The stand-alone version has been flawless. (Visit www.arturia.com for details on compatibility with your system).

To be honest, I was ready to buy just the physical MIDI controller for $300 – it’s that nice – and, yes, you can use it to control any MIDI software. When you include the perfectly integrated software synth in this package, it’s a pretty insane deal. Go download the demo and hear it for yourself, but don’t underestimate how important the integration of the hardware and software are in making this hybrid synth a winner. This design gives me a lot of hope for the possibility of having increasingly soulful connections to modeled analog instruments. Allen Farmelo www.farmelo.com

This review originally appeared in Tape Op and was co-written with Jessica Thompson.

In a digital system, the clock signal is used to generate a common time reference for the flow of data in the system. The clock signal goes to all the elements that need it and basically regulates the flow of zeros and ones so that everything works together synchronously. Clocks are a hot topic right now, with debates showing up on gearslutz.com, here in Tape-Op (see last issues letter section), and generally among record makers. Why so hotly debated? Because clocks can have a highly significant impact on sound, yet there is no solid consensus as to why that is. Regardless of the hard math and science behind digital clocking, the aesthetic impact of different clocking systems remains both subjective and context dependent, as with any piece of gear in our racks. So as reviewers, we have chosen to sidestep the technological debates as much as possible, and move forward with two simple assumptions: 1) different clocks make the same recordings on the same system sound different; and 2) in some cases, that difference is big enough to convince one to buy a new clock.

The heart of the Isochrone OCX ($1500 street) is a temperature-controlled oven housing a discrete transistor crystal oscillator. This translates to higher stability and four to ten times lower jitter than competitors, according to Antelope. It supports sample rates from 32 to 192khz, and – this is a bonus – it is capable of outputting multiple sample rates simultaneously (thought we didn’t use this feature). Take a look at its back panel, and you’ll find eight word clock outputs, two AES/EBU, two S/PDIF outputs, as well as compatability with Digidesign’s 256x Superclock. This sleek, 1RU silver box could easily be the classiest looking piece of gear in your rack. One design element we particularly liked is the prominent red LED read-out of the sample rate. It’s front-and-center enough to remind you which sample rate you’re clocked to, which we suspect will prevent the occasional clocking mishap[.] (Allen confesses to having mixed a song tracked at 48khz at the slower and lower-pitched 44.1khz for about twenty minutes before catching on).

The Isochorne 10M Rubidium Atomic Clock, the OCX’s sleeker, more expensive 2RU companion ($6000 street), is designed to enhance the OCX with atomic clocking technology. When interfaced with the 10M, the OCX switches from its crystal oscillator to the 10M’s Rubidium core. Basically, the rubidium element disciplines the crystal to its hyperfine oscillation (over 6.8 billion Hrz), which produces 100,000 times better accuracy than your Swiss-made quartz-driven Rolex. We’re talking a deviation of one second in 1000 years. This is the same technology used for GPS and broadcast clocks. (Just in case you’re worried, the manual assures us it’s not actually radioactive).

In our test of the OCX and the 10M, we first used The Farm (Allen’s mix room in Brooklyn), a system that relies on the clock to regulate the flow of data among a handful of digital units. Here’s the setup: Pro Tools HD interfaced directly from the Core Card to a Lynx Aurora 8 converter, which is connected digitally to both a Cranesong HEDD 192 for A-D-A conversion to a stereo analog mix-buss insert, and to a Dangerous Audio D-Box for monitoring. By connecting the OCX to the word clock input on the HEDD, we were able to switch the system’s master clock between the Lynx (Pro Tools sees it as the internal clock), the HEDD (Pro Tools sees that as the external Word Clock), the OCX (by telling the HEDD to use its external Word Clock input) and the OCX with the 10M attached. The D-Box uses its digital input as its clock source, so it conformed to whatever master clock we assigned. If you’re still reading, what this all means is that we could easily switch between four different clock sources and all the digital gear would conform to whichever clock we selected.

We threw up a number of different mixes, and our first impression was: “hey, different clocks really sound different.” Both the HEDD and the OCX seemed to deliver a similar amount of information and fidelity, but the HEDD had a stronger center, while the OCX offered a slightly wider stereo field. One way to describe it is that the HEDD presented a convex soundstage and the OCX a concave soundstage – two rather different shapes holding about the same amount of information. The HEDD also seemed to have a bit more midrange presence, while the OCX was a little more elegantly detailed in the highs. In less abstract terms, the HEDD rocked out with a bit more sonic glue, and the OCX was a little more elegant and spacious. The Lynx clock didn’t reveal as much detail, especially in the airy region, but also in the deeper lows (and Allen always clocks his HD system externally to the HEDD for this reason). On the whole, the OCX would be an excellent choice to anyone looking to find an external clock to improve the sound of any digital system with internal clocks that might be worth improving.

Where the OCX really showed its stripes was on a Digi 002 system, belonging to Brooklyn-based engineer, Matthew Agoglia. Matt’s room is a great example of a “real world” mixing and tracking room: Digi 002 running through a Hafler power amp into Yamaha NS-10Ms that were awaiting new woofers. From within Pro Tools LE, we put up Emmylou Harris’ song “Deeper Well” off of the Daniel Lanois produced album Wrecking Ball, a track filled with endless sonic details and effects tumbling around in the background. The difference between the 002 and the 002 clocked to the OCX was absolutely revelatory! There were elements in the tracks that simply didn’t make it to the speakers without the OCX. We listened to a lot of stuff and found the same thing over and over. It’s hard to imagine a single purchase that would upgrade a system in this realm so significantly and pervasively. Everything one does on this system – tracking, monitoring, mixing, printing, bouncing – is going to be significantly improved.

Back in Allen’s studio we hooked up the 10M to the OCX (a simple BNC patch), threw up one of Allen’s mixes, and the whole world changed. It felt like there were about five extra spaces in the stereo field where one might have placed an element of the mix, and elements we hadn’t heard before were plain as day. Things like acoustic guitar finger squeaks, the singer’s moist mouth mutterings, more of a ride cymbal’s over- and under-tones, aspects of a kick drum’s raspy attack, reverb tails, tape-echo trails, and even compression artifacts were showing up, seemingly from out of thin air. The soundstage gained a depth that seemed to reduce masking between elements that occupied the same frequency range, as if they instinctively found space in front of and behind each other based on how wet or dry they were. Apparent loudness went up a notch, without changing the volume of anything (a psychoacoustic phenomenon? – we’re not sure), and there was a noticeable low-end extension. Beyond the details, the whole of the parts was a total pleasure to listen to, and evoked a far more vivid image of all the aspects of whatever mix we put up. The10M just made the music far more engaging and emotional (and only made the current MP3 paradigm seem more criminal).

In fact, with everything we put up the 10M was a mind-blower, but on one track we actually found ourselves more interested in the lyrics. Fascinating. A stripped-down, crawling version of Neil Young’s “Harvest Moon” by the nomadic singer Jess Lee with Allen backing on a simple organ part was rendered in such detail with the 10M that individual harmonic overtones in Jess’ voice (tracked with a SM58) almost seemed like individual sonic elements. The organ (run through a vintage RCA tube PA into a Senheizer 421) fanned from one warm shade of orange into a complex spectrum of warm, burnished tones, and previously subtle oscillations became rhythmic pulses Allen hadn’t heard so clearly since tracking it. Somehow, these details drew both of us far more deeply into the story Mr. Young weaves in his lyrics, and into Jess’ lonesome interpretation. The recording took on a vitality and intimacy that was, it seems, hiding somewhere in the digital code.

So, can a clock make a difference? Ha! – especially when you’ve got a sensitive mixer/producer and a discriminating mastering engineer geeking out on a really nice system in a well-treated room. But what about the so-called real world? Is the clock going to help a file weezing its weary way through the world’s worst D-A converter and a pair of 10-cent laptop speakers? We printed mixes from the four different clocks to find out, and in a blind test we were able to hear differences on a laptop, for sure. However, the differences were certainly diminished by the limitations of the playback system – if you can even call a laptop a playback system. But we don’t work our butts off to make laptops bring people to tears; we do it so that no matter where a recording ends up, it has the best chance of being rendered in all its intended qualities. And, as we look toward brighter days when MP3s have gone the way of the Edison Cylinder, there is no time like the present to consider tools that will generate zeros and ones that will outlive the current lo-fi trends and shine like diamonds in the high-fidelity renaissance of the future.

Whether you’re on a pro-sumer system and can use the OCX to bump up your rig, or you’re a world-class mastering engineer or mixer who can afford the 10M, both of these clocks are capable of making a big enough difference to warrant serious consideration of a purchase. In the case of the discerning, high-level professional, it’s apparent that the margin for sonic improvement is often pretty narrow, yet we assure you that trying the 10M is worth it. As we said, we’ll leave the technological arguments to those with the minds for it, but if you’re like us and want to do all you can to render human musical performances with as much depth, dimension, detail and love as possible, give the Antelope clocks a listen and hear for yourself what they can do for your recordings.

This review originally appeared in Tape Op.

The emulation of analog gear in the digital realm is nothing new, but the emulation of analog gear in the analog realm is a new concept—analog modeling. AnaMod was founded in 2006 by two long-time industry innovators, Dave Amels (Bomb Factory, Voce, Tape Op #31) and Greg Gualtieri (Pendulum Audio, Tape Op #38). AnaMod products are entirely analog and do not process audio in the digital domain, yet the same patented mathematical modeling found in digital plug-ins for Bomb Factory has been applied to the creation of AnaMod’s all-analog products. The company’s trademarked (and also patented) AnaMod Process takes a mathematical model and implements it using analog building blocks. At first glance, it may seem like the company is building recreations of vintage gear, but when you realize that they’ve fit their version of a Fairchild tube compressor into a single 500-series solid-state module (the recently released AM660), the AnaMod concept begins to stand out. Additionally, there’s no D-A or A-D conversion, no latency, no hidden fees for software upgrades, and no system incompatibility limitations. Hardware emulating hardware—an interesting concept, indeed.

Now imagine that you’re mixing a record, and there is a machine room with four tape-decks, say a 3M M79, a Studer A800, an Ampex 351 and an ATR 102, all running like tops. Also in the room is the industry’s best tape-op waiting to throw on fresh reels of GP9, 456 or new-old-stock Scotch 111, transfer your tracks to the fresh reel, and calibrate the tape decks instantly to taste. Let’s also include instant re-biasing, the ability to completely remove or add as much hiss as you want, and miraculously, the ability to hear your mix without the coloration of the tape or the deck (hard bypass). Welcome to AnaMod’s ATS 1 Analog Tape Simulator.

The ATS 1 is a 2RU-height stereo unit with the look and feel of an old Ampex 350, sporting a silver front, black vintage-style knobs, two well-lit VU output meters, and two illuminated, square record and stop buttons that engage and bypass the unit, respectively. Large input and output knobs for each channel are the only individual L/R controls, leaving the rest to affect both channels simultaneously. Some of the controls correspond directly to the settings and alignment pots you find on a real tape deck: Bias level; HF Repro level (a treble boost/cut on the playback circuit); LF Record level (a bass boost/cut on the record circuit); Tape Speed (7.5, 15, and 30 IPS); and a meter calibration switch, allowing the user to go from 0 dB (185 nWb/m), up to +12 dB in 3 dB steps, effectively simulating the standard “overs” at which we often calibrate for different tape formulas and/or desired coloration and compression. There’s also a knob to add Hiss (from none to way too much). Two rotary switches select from up to four tape machine and four tape formula models. Each model is a user-installable “personality card” that fits into a slot inside the unit. Doing the math, there are 48 different selectable combinations (4 decks x 4 tapes x 3 speeds), and those can all be infinitely tweaked with the other controls. The ATS 1 is certainly the most complex and thorough tape-emulation system out there to date, offering an enormous range of subtle shades of harmonic complexity.

Given the vastness of possibilities, it would be nearly impossible to test the accuracy of each model combination. Besides, simulations always sound a bit different than the real thing, and when you’re dealing with vintage gear, especially old tape-decks that have been repeatedly serviced, running relatively inconsistent tape formulas (some of which have not been manufactured in decades), you’re never going to have consistency from one machine to another, or even on the same machine from one day to the next. So why worry too much about consistency from real to simulated? However, let me quickly say that the ATS 1 really sounds and behaves like a tape deck. When you push the inputs, the low-end builds up and the dynamic range begins to shrink, eventually giving way to a dark, crunchy distortion. When you slow down the deck, the characteristic bump in the low end and a slight loss of air ensue. When you under-bias the tape, you hear sibilance in the highs and fuzzy harmonic distortion in the lows. Because you’re controlling parameters that are named after—and behave like—those you’d find on a tape deck, you end up thinking within the analog tape paradigm, something I don’t seem to get to do enough these days. (If only the ATS 1 could also recreate the more humane pacing of a session with wait-times for the transport to rewind and the tapes to be changed.)

You certainly don’t need knowledge of analog tape technology to get usable sounds out of the ATS 1, but such knowledge gives one insight into what these controls are doing within the modeled circuit and helped me feel confident operating it right away. Reversing that logic, the ATS 1 could be an excellent place to begin to learn analog tape technology. What does going +6 on an A800 running GP9 sound like at 15 IPS? Turn a few knobs and—more or less—find out!

While mixing in the box (Pro Tools HD is my box), one of the most important things I do is to run my entire mix through an analog 2-bus chain in order to add harmonic complexity, girth, depth, width, mojo and vibe. It’s not a revolutionary technique at this point, but one that begs for constant refinement and expansion. Depending on the material, I use many combinations of analog gear, but my typical chain is a Cranesong HEDD 192 D-A converter to 1073s (either Vintech or real Neve, depending on where I am) to an API 2500 stereo bus compressor and back into the HEDD for tape and tube simulation and A-D conversion back into Pro Tools. I think of it this way: Pro Tools is my multitrack, often accentuated as such with Cranesong Phoenix, Massey Tape-Head, and other “color” plug-ins; the 1073s and the API (or whichever line-driver and compressor) give me my console coloration; and the HEDD’s tube and tape emulation is my 2-track deck. Replacing the HEDD’s processing with the ATS 1 as my simulated 2-track machine was a great way to get to know the unit, and I imagine, one of the more common ways people will use it.

On a mix I recently did for NYC-based musician and singer Jonah Smith, I was working on a slower, soulful tune featuring bass, drums, Wurlitzer piano, electric guitars, effected guitar loops, horns, strings, lead vocals, and a full back-up vocal section (all beautifully produced by the venerable Malcolm Burn). I was very happy with my mix using the HEDD processing to emulate my 2-track deck, yet was curious to hear the ATS 1 at work on this lovely, complex mix. As a general rundown, here’s what I heard at 30 IPS with the bias and EQs set neutrally: the M79 deck made the low-mids punch and softened the top end; the A800 deck gave me a slightly more open top-end and focused the low-end; the 351 was lush, deep, and even fuzzy (probably a result of emulating tubes) and sounded most like the HEDD’s pentode and tape emulators combined; the A102 produced a narrower image overall and gave the mids a nice smoothness. I can’t get into all of the combinations with the three tape formulas. However, the general vibe is that the GP9 simulation is the most hi-fi with a tighter bottom; 456 sounded fatter on the bottom, smooth in the highs, and a bit compressed; and 111 was more closed on top and the most compressed overall. On this mix, my favorite was the 351 running GP9, which gave the horns and strings an added sense of depth, while adding a glowing aura to the whole soundfield.

Now, imagine trying all this again at 15 IPS, now 7.5 IPS, now while driving the inputs a bit for more compression and coloration, now while varying the bias, and now while adding and subtracting treble and bass from the repro and record circuits respectively. It’s an immensely vast palette of tape-style subtleties.

Next I strapped the ATS 1 to a soloed drum submix and found myself leaning toward 15 IPS, Studer A800, and 111 tape for a really punchy kick drum, a softening of the cymbals, and a nice forwardness in the snare. It was nice to know that, should this be too dark in the overall mix, I could change the setting quickly to make things sparkle, perhaps moving to GP9 at 30 IPS.

Want to get bass to growl a bit? Try running Scotch 111 on the Ampex 351 at 7.5 IPS. Nice. In fact, there’s a whole world of tonal shaping to be done on the 7.5 IPS setting, which, to my ears, brought the most color out of all the different deck and tape models.

After about a week of mixing with the ATS 1, I noticed my thinking and vocabulary shifting to the paradigm of analog tape. This was the first time in my Pro Tools mix room that I’d begun to think in terms of IPS, biasing, or “running at +6″, and it was certainly the first time I’d thought things like, “let’s switch it over to GP9 and see how it sounds”, or “throw it on the 351 and give it a whirl”. I always welcome a paradigm shift, and found this aspect of using the ATS 1 in a digital studio really refreshing.

In the days when all stages of recording were, by necessity, done on tape, it was standard practice to hit tape during tracking, hit it again during a bounce or two, and then hit the 2-track tape during mixing. That layering of harmonic complexity helped form the sound of what are generally regarded as the hi-fidelity masterpieces of modern recording (fill in your favorite example here). Using the ATS-1 to build harmonic complexity throughout the various stages of recording, starting with tracking, can help one create more distinctively colored recordings in the digital realm. I found that an electric bass recorded through the ATS-1 sat way better with a drum submix that was run though the same settings, and that hitting both of those again with the ATS-1 during mixdown really glued the rhythm section together. Gradually building up harmonic saturation always seems to work better for me than hitting any track just once. As my mother would say: “It’s cold out there, so layer up!”

Similarly, when working in many different studios, tracking with the ATS-1 might help the project gain a sonic consistency it might not otherwise have. Think of it as tracking to the same reel of tape for each session. No matter where you are, the sound of that reel will help define the sound of the record. And, toting a 2RU unit is far easier than lugging multiple reels of tape.

Unexpectedly, perhaps the most interesting features on the ATS 1 are the bias and hiss knobs. Bias (more specifically AC bias) is basically a high-frequency signal (generally around 100 kHz but as high as 423 kHz for the ATR 100 series) added to the signal going to tape that activates the magnetic particles to improve the linearity of the medium, and subsequently, the fidelity. If you don’t understand bias, don’t worry; there are many—including me—who consider the physics behind it a mystery. But the sound of bias on tape is not a mystery, and those who work with analog tape know it’s an essential aspect of the technology that dramatically improves frequency response, reduces distortion, and increases signal-to-noise ratio when set properly.

Whatever the AnaMod team did in the creation of the bias control, they created something truly unique in today’s market and could easily package the bias and hiss controls as a separate stereo unit (if that’s even possible without the deck and tape models running). Why should they do this? Because these two knobs can change the nature of the program material in ways I just don’t think you can get from any other piece of gear, aside from an actual tape machine (where hiss is not optional!). Let’s put it this way: “biasing” a digital mix is a trip.

On a mix of thirteen horn players (bones, saxes, tuba, bass clarinet, trumpets) improvising a somber, tumbling dirge of “Amazing Grace”, hiss was absolutely useful and lovely. Pushed too far, it was just hiss, but bled in carefully, it added a lovely layer of sonic gauze that gently filled the dark spaces between the widely panned sections. Perhaps this is nothing more than associative conditioning on my part—just me hearing hiss as a part of the world I expect free jazz and “Amazing Grace” to occupy. However, by over-biasing a bit, I was able to cramp the frequency response just enough to reign the hiss back in. The result is hard to describe, but I’d say I was able to unite the different voices without losing any of their individual clarity. In my book, that’s an accomplishment, and something mastering engineers in particular might want to check out as an alternative noise-shaping (dithering) method. Bias and hiss are certainly part of that “analog magic” we are prone to love.

Along similar lines, the bias and hiss controls can be used together as an untraditional but oddly effective noise-reduction scheme. On a mix for a solo electric guitar performance (think Eno and Glass playing for a Transcendental Guru), the guitar was DI’ed from a Line 6 Echo Pro into a 1073 and then a Requisite L2M tube limiter. Nice chain—except that the echo unit introduced some noise with which I wasn’t happy once I was bringing things up to broadcast (post-mastered) level. Again, by adding a subtle amount of hiss and really over-biasing the Studer A80 running GP9 at 30 IPS, I had one of the most effective noise-reduction units I’ve ever used.

How does the ATS 1 stack up against my Cranesong HEDD, the Cranesong Phoenix plug-in, and the Massey Tape-Head plug-in? In overall character, it’s closest to Phoenix, creating a smooth, subtle coloration of the signal, and with increasing compression and coloration artifacts appearing as you drive the unit harder. With Phoenix’s five different flavors and three different colors, these two are the closest. Compared to the HEDD, the ATS 1 is far more versatile, but generates a very different kind of color. The HEDD’s tube emulations generate depth in the soundfield that only the 351 model came close to, and I think this points to the accuracy of the modeling of the ATS 1 (and the HEDD). The ATS 1 is furthest in sound from Massey Tape-Head, a plug-in I love to death for its ability to wake up a dead guitar or snare, for example. All told, it’s a real treat to have all these different processors on hand to add different types of harmonic complexity to whatever I’m working on, but only the ATS 1 had me thinking and feeling like I was actually working with real tape.

In comparisons with recordings actually made to tape, I was limited to work I’ve done on an Otari MTR 90 2” with 16-track heads calibrated at +4 running GP9 at 15 IPS, and mixes run to a Studer A80 1/4” 2-track running GP9 at 30 IPS, so there are obvious obstacles in the way of an apples-to-apples comparison here. But listening to the tape recordings against the ATS 1 only furthered my belief that AnaMod is onto something with their simulation methods, as the general vibe is very similar. On a Billy Nayer Show tune I recorded and mixed to the Studer 2-track and to Pro Tools, I got very close to emulating the analog sound using the ATS 1’s Studer A800 model, but actually appreciated what some of the other combinations were doing better. Again, for me, this box is about having that magic machine room with all those decks and reels. (Also, don’t underestimate the importance of adding hiss when trying to match a digital recording to a tape!)

I want to stress that the sonic impact of the ATS 1 (and of tape in general) is subtle, even hard for the inexperienced ear to hear. Hearing the difference between the tape formulas, and how they react to being driven differently, is something akin to tasting wine and being able to identify the wood of the aging barrels. At $2995 list, the ATS 1 might be a hard sell for those who can’t taste the wood, so to speak.

However, for those who grasp and desire the power to impact the subconscious of the listener with subtle colorations of sound, I’ll say that the ATS 1 is currently the most versatile and powerful piece of kit on the market in this category, with just about no competition. And I can’t stress enough how important a box like this can be to the professional mixer working in the box, especially if you’re missing the vibe of tape. I would also expect that mastering engineers are going to love the ATS 1, especially when there’s that album tracked to tape except for the one Mbox home-job, or when working a re-mastering job where the sound of a particular deck would show respect for the original, or when challenged with sonic inconsistencies across a diversely produced record. I can think of more than one mastering session where the ATS 1 would have been really helpful in pulling things together. Moreover, for mastering, we’re dealing with hardware, which is easily integrated into the facility and eliminates the need for cross-platform compatibility. Mastering engineers should really check out the ATS 1.

My hat is off to the AnaMod team for taking a truly novel approach to gear design and opening up a whole new field of development in recording gear. I look forward to new personality cards for the ATS 1 and to more analog models in the future. Bring it on! ($2995 MSRP; www.anamodaudio.com)
–Allen Farmelo, www.farmelo.com

This review originally appeared in Tape Op.

Ever since the studio where I do my tracking (Mavericks in NYC) acquired a pair of vintage RCA BK-5A ribbon mics, I’ve become a big fan and have found a ton of uses for them. It’s difficult to track down the production dates of RCA equipment, but the BK-5A was likely produced from roughly the mid 1950s into the later 60s (there are photos of a very young Johnny Cash, circa 1957, singing into one, and the DJ in The Carpenters’ 1976 video for “Calling Occupants of Interplanetary Craft” is using one). The BK-5A looks like a Revolutionary War cannon with art-deco details, but it’s unique qualities are more than skin-deep, as it has a hyper-cardiod polar pattern (most ribbons are figure 8), a frequency range of 50 to 15,000 hertz, and was designed to withstand SPLs caused by gunshots during broadcast and film work. RCA did some interesting things with the chambers behind the ribbon to accomplish this ruggedness, and it seems to me that the challenge of designing a mic that can take a bullet has resulted in it’s unique sound. There is also a three-position low-end roll-off switch (M, V1, V2), and the BK-5A shipped with a tennis-ball sized, perforated-metal windscreen that, according to Bob Crowley of Crowley & Tripp, doesn’t really do anything. Bob also told me that the BK-5A shares its transformer and a few other bits and pieces with the formidable RCA 77DX ribbon mic, and that, in his opinion, the BK-5A sounds similar to the 77DX in unidirectional mode.

What I love most about ribbon mics is their ability to deliver detailed imaging in the upper mids, those critical frequencies where so much of a recording’s vital characteristics are. The BK-5A focuses on the same sonic area that a SM57 seems to focus, but the BK-5A captures details in that range that give it a transparency, realism and detail I rarely get out of a dynamic mic. Anyone who has used a Royer 121, for example, can attest to the detailed realism and imaging a ribbon can deliver – now imagine that without all the velvety lows and silky highs of the Royer, and you’re getting the idea of what the BK-5A sounds like. Or, imagine an SM57 that doesn’t get nasal or snotty, but instead delivers 3D imaging and nuanced details – to my ears, that’s the BK-5A.

For me, this is a thoroughly useful set of characteristics, since during mix-down I am often pulling out bottom end on tracks to make room for the bass, kick, or whatever is carrying the low end of the music. I also really like the lack of shimmering, sizzling highs for similar reasons, as tracks I’ve recorded with the BK-5A seldom compete with important high-frequency content. So, I’ve been using the BK-5A for tracks that I want to occupy the middle of the mix without getting in the way of the highs and lows.

For example, the nuances captured on violin were particularly impressive, with rosin, horsehair and wood rendered in 3D, without any irritating upper harmonics. This violin sat right in the mix as I wanted it to. And I love the BK-5A as a room mic on drums; it captures all the sound I want, without competing with the overheads or the kick. The snare just punches through, giving me ambience where I wanted it. On a record I’m currently mixing, I’ve taken to squashing the BK-5A room mic with a compressor so it breathes with the beat. This technique creates a really useful sustained snare sound and extra action from the cymbals, but without the sloppy washed out sound that I sometimes get when using condensers in the same way. Throwing the BK-5A over the shoulder of the drummer gave me a crunchy, vintage sounding, lo-fi mono drum track, but didn’t obscure any nuances of the drummers performance.

I found another very interesting use for the BK-5A. In the vocal mic shoot-out I did with NYC-based singer-songwriter Sarah Tolar, Sarah was giving killer performances through the BK-5A, but in the context of her warm, intimate record (lots of Steinway and upright bass), the BK-5A made Sarah sound like she was coming through a transistor AM radio. When she sang through the Royer 121 the tone was rich, detailed and sublimely warm, but her performance sounded disengaged in comparison. Sarah and I talked about it, and something in the mid-range emphasis of the BK-5A was giving her information she just couldn’t get from other mics that have more full-range sound. She is also a seasoned stage performer, quite used to SM58s and others mics of that ilk. In this case, I hung both mics, had Sarah monitor through the BK-5A, and then used the Royer for the record – a win-win situation that worked for the entire album. (If you try this with two ribbons, watch out as the magnets like to pull at each other. I don’t know if this affects the performance of the ribbons, but I didn’t hear anything.)

I’ve heard male vocals recorded with the BK-5A that work beautifully; it’s ideal for a somewhat eerie, mid-rangy sound when something more lush isn’t appropriate – great for a quasi-lo-fi vibe. In fact, anytime you want a limited frequency response, but don’t want to sacrifice detailed imaging, the BK-5A is a great mic.

I would never recommend the BK-5A to anyone buying his or her first mic, or to anyone rounding out a small mic collection, but I would tell anyone looking to add another color to an already well-rounded mic collection to try to get one. Buying a BK-5A is a bit of work, as you’ll have to search eBay or another source, and possibly have the ribbon replaced. I should also mention that the later model, the BK-5B is, as Bob Crowley told me, only different in that it has a slightly altered vent behind the ribbon. Though the BK-5B literature listed a far wider frequency response, Bob feels there is no sonic difference between the BK-5A and 5B. Prices on eBay seem to hover somewhere between $750 and $1000. Allen Farmelo (www.farmelo.com)

This review originally appeared in Tape Op Magazine.

A number of years back I was tracking a record in a room with a 24-fader Digidesign Pro Control, and then migrated to a situation with only a keyboard and a mouse. About half way through the first day without the Pro Control, I turned to my esteemed colleagues and said, “Man, I wish I had just one fucking fader!” Frustration, it appears, is the real mother of invention, and we all lit up, wondering why this seemingly obvious piece of kit wasn’t available. Everyone I shared the idea with said it was a brilliant idea and that I should pursue it. The product name One Fucking Fader was, of course, a marketing rocket ship just waiting for ignition, though I soon cooled my jets and settled on The Monorail. In all truth, I did pursue patenting The Monorail, a name I still hold the rights to, only to learn a lot about inventing and patent law, including some nonsense about “not a novel assemblage of prior technologies,” and, “it is not recommended that applicant pursue patent.” I was actually relieved. It’s an unthinkable load of time, talent, and treasure to manage just an idea for a product, and I have since developed a really deep respect and gratitude for all the designers and manufacturers of the gear we all use every day making records.

I felt vindicated when Presonus released their single-fader control surface, the Fader Port (reviewed in Tape Op #61), and doubly so when Frontier Audio followed up with their AlphaTrack. I tried the Presonus out, and was kind of bummed, to be honest. It wasn’t very sleek, it required a wall wart, the name wasn’t that cool, there wasn’t a display to indicate which track or parameter was selected, and the buttons were kind of hard to push. When the Alpha Track showed up on the market, I was pleased to find that it looked pretty mod (think Star-Trek), was powered via its USB connection, and included a handsome blue LCD display. The name? – well that’s really not all that important.

AlphaTrack includes a 100mm, touch-sensitive, motorized fader with true 10-bit resolution, three touch-sensitive encoders that provide quick control of pans, sends, EQ, plug-ins, and automation. The 32-character backlit display shows info about what you’re doing with the encoders, and it’s a huge advantage because there are so many parameters available for control, and most can be flipped with the fader, too. At the bottom is a touch-sensitive jog and shuttle strip, something like a narrow track-pad you’d find on a laptop. The strip allows you to scroll through a project’s timeline, vary shuttle speed, and navigate through markers. It takes a minute to learn the commands (a combination of one and two finger moves), but, once I did, using the shuttle strip was quite intuitive. AlphaTrack has 22 buttons and 21 LED’s, including transport controls, track record, solo and mute buttons, and automation-mode indicators. On the back is a footswitch jack for punching-in, a sweet feature when recording yourself.

AlphaTrack connects to your computer with a USB cable, requires that you install a small program to get it running, and that you make it known to your DAW as a HUI device. It works on both Windows and Mac, though I’ve only installed the driver on a MacPro and run the device from within Pro Tools (for more info on compatibility, go to www.frontierdesign.com). It was fast and easy to have it up and running, and as soon as I did I was writing vocal automation with a fader within minutes. I finally had my one f’n fader.

Working with AlphaTrack is really fun, intuitive, simple and effective. For me, it’s been a liberating tool for writing automation while mixing in the box, especially across a lead vocal track. Compared to using a mouse, I’m not only able to get the intuitive feel of a fader, but I can turn off the damn screen! I have a hot corner on my monitor that turns on my favorite lava-lamp-esque screensaver, and I toss the cursor in that corner whenever I can. On an average day of mixing with the AlphaTrack, I’d say that the screen saver is on about 30% more of the time, maybe more. That’s huge. I love Pro Tools, but if I never look at it again, I’ll be ok. If you’re like me, something different happens when visually disconnected from a DAWs interface. I can literally feel a different part of my brain begin to work – the listening part. It’s the way I feel when I work on tape, and I love it. It’s also why many of us close our eyes when we taste something delicious, to heighten one sensation by cutting out another. Writing automation in touch mode with the AlphaTrack allows me to use the transport controls and the shuttle strip to flexibly navigate my session with my screensaver on, and use the fader, and my ears, to do my rides. Awesome!

Of course, there’s a whole lot more you can do with the AlphaTrack, and each user will find a way to make it suit the demands of her or his situation. It’s quite adaptable and assignable (again, read up at www.frontieraudio.com for more info on that).

Having designed The Monorail to some degree, I have some ideas to throw out into the creative ether. The AlphaTrack isn’t dinky, and might be a bit big for use on an airplane tray table, for example. What about a low profile and very narrow version with fewer controls for ultra portability and elegance? Make those daisy-chainable (snap them together like Legos), and build a modular control surface that grows with your needs. I also envision a version that would mount in either a 60mm or 100mm console-fader frame. Drop that in next to the master fader, and have a DAW-integrated fader right in your board. Daisy chains those with a jumper cable and build out as needed (I envision these in the sometimes unused space beneath a patch bay, perhaps). Gear makers, I’m ready to be vindicated again.

In the meantime, I aggressively advocate the AlphaTrack to anyone working without a control surface. At $200.00 street, it’s a steal. Allen Farmelo www.farmelo.com

This review originally appeared in Tape Op.

There aren’t a lot of de-essers out there, probably because they’re not very sexy and not at the top of many people’s wish-lists. But a good de-esser is invaluable, especially now that so many people are tracking vocals at home through crappy converters with harsh condenser mics. I hate to look down my nose at artist-empowering, cost-effective recording equipment, but consumer grade vocal chains produce some of the ugliest fricatives I’ve ever heard or seen! Man, a cheap condenser through a weak pre and converter will leave an ess with a big, ugly angular waveform you’d expect from a buzz-saw, not the human mouth. And then, because the converter stinks, that nasty waveform isn’t translated in the monitoring chain, so no-one hears it until it’s played back on a better system, often during mixing or, worse, during mastering. It’s here, on the better monitoring path, that the soft, sensual human mouth reveals itself as a harsh, hacking buzz-saw. At this point, a powerful, versatile, de-esser with automation starts looking pretty sexy.

In steps Massey with a de-esser plug-in ready to take the prize in its category. The main controls include a large reduction fader, a frequency knob, a band-split button (which engages something like a cross-over that leaves the lower frequencies un-processed), and a dry…wet knob (basically a ratio control). All said, the main controls are very easy to use, do what they’re supposed to do, and are consistent in their rate of change. The reduction slider is big enough to make small changes easy to automate, and I appreciate this design feature a lot, as I can be a little less cross-eyed while de-essing during the final hours of a mix. The high-resolution, LED-style output and reduction meters (think DBX here) confirm visually what you’re hearing, and seem to be quite accurate representations of the sonic behavior of the plug-in.

At the bottom of the plug-in is a collapsible panel with more controls. The response switch changes the behavior of the dynamic threshold used in the de-esser; it acts something like a knee control on a compressor in that it changes the aggressiveness of the curve. The speed control alters the reaction time, which is easily heard and seen on the meter. Then there is the re-ess option, which inverts the operation of the de-esser, boosting sibilance, rather than reducing. (Weird, right? More on this in a bit). Finally, there’s an output trim, which is always useful, especially when you start messing with the re-ess function.

Then there is the listen section, which, on a de-esser, can be as valuable as the processing itself in getting the behavior dialed in properly. The standard listen function on most de-essers is called the filtered mode here, basically playing just the portion of the signal feeding the sidechain (that is, you hear what the de-esser is reacting to). In the invert mode, however, you mostly only hear signal when the de-esser is working. In invert mode, it becomes obvious when the processing is working, and how hard it’s working. You can quickly tell if the processing is killing the vocal and easily find those sections where automation may be necessary. Lastly, you can choose to be in listen mode continuously (On), to turn it off all together (Off), or to have listen mode come on automatically whenever you touch the frequency knob (Auto). Each of these settings is useful, depending on how you’re trying to hear the audio you’re processing.

Clearly, this plug-in is built to be a versatile and powerful tool, and it is. In my use, I was first and foremost impressed by how easily I could dial in which frequency to select for processing vocals while mixing. I attribute this ease to the sonic insight of the invert listening mode and to the visual clarity of the reduction meter. Together, I could hear and see the behavior of the de-esser very clearly. On one particularly buzz-sawing male vocal (tracked through a decent API pre, but with an unflattering condenser mic and a mediocre converter and clock), I ended up automating the De:Esser extensively. Again, that invert listening mode really helped move this rather dull process along – much faster than other de-essers. On another record, I tracked male vocal, classical guitar and piano. The U87 I used on the vocal was a bit too crisp for the sonic context, and I needed general de-essing throughout. (Note to self: remember to try more ribbon mics when recording vocals digitally into music without cymbals.) During this mix, I ended up layering two instances of the Massey De:Esser, one set around 4k, and the other at around 12k, each reducing lightly, and got great results. In this case, by putting the 12k de-esser first, and setting it in band-split mode, I was able to let the lower frequencies pass through un-processed, so that the next instance of the de-esser set at 4k caught the more meaty fricatives. I can’t emphasize enough how important the band-split mode is to this cascading technique, basically creating a much more selective and nuanced de-essing processor.

Because a de-esser is really a band-dependent compressor, they’re very useful – even fun – on other sources. De-essers are commonly used on acoustic guitars whose pick attack is a bit too much, and on drums with harsh cymbals. I encourage anyone to play around with the Massey De:Esser in these kinds of situations, especially on drum over-heads, where I had very good results dealing with some bright, tight hi-hats played with nylon-tip sticks.

But, the most fun I had with the the De:Esser was using the re-ess setting on drums. This was a 3-mic technique, with a Coles 4038 above the kit, an U87 about 10 feet out in the room, and a D112 on the kick. The Coles is about as dark as a mic gets, while the U87 is one of the brightest. I love this combination, as I get a very lively room sound and a tight, punchy sound from the overhead. However, come mix time, I sometimes want more highs out of the Coles over-head, just to get some more sparkle. Using the Massey De:Esser in re-ess mode produced some very interesting results, as I was getting a dynamic high-end boost. On louder snare hits, for example, more highs would spill out. It was almost like having a bottom-snare mic with a fast-opening envelope on it. (Again: Weird right?). Setting the response switch to “D” (designed for drums), made the detector fast, and allowed the drums to breathe in a very unique, and usable way. Here’s where the dry…wet knob came in handy, allowing me to dial in as much of this effect as I wanted. I imagine the re-ess mode to be one of those tools that could become a secret weapon in certain situations, or when looking for something different to give a mix a little lift.

A flexible de-esser is an essential tool for mixers at all levels, especially in the digital realm. Like I said, de-essing ain’t sexy, but an affordable, versatile tool that you might end up using everyday has some serious appeal. For $89, this plug-in is a slamming deal, and you’re buying directly from an accessible, individual entrepreneur who really gives a shit about the end users of his products. Download the free trial (no interruptions to audio – thank you very much) and take it for a spin. I also encourage you to take some time getting used to it; it’s a deeper plug-in than you might expect from a de-esser, and deserves some serious investigation. Available as direct download from www.masseyplugins.com as TDM (up to 96k) and RTAS (up to 192k) for Pro Tools 7. Allen Farmelo www.farmelo.com

This review originally appeared in Tape Op #61.

In Issue #60, I gave a glowing review of the Focal Solo6 Be studio monitors, and now I turn to the Sub6 Be, a mighty subwoofer especially designed for use with Focal’s 6-range speakers. The Sub6 Be arrived about a week after the Solo6s, so I’d had time to acclimate to the bass response of the Solo6s before taking the plunge into the deep end. With a 350 watt BASH amp powering an 11-inch W-Cone driver in a cabinet including a large section laminar port, the Sub 6 has a frequency response reaching down to 30Hz. The Sub6 is designed to serve as the LFE channel speaker in a surround system or as a bass or sub-bass complement in a stereo system (2.1 or 2.2). In my use, it was as the subwoofer in a 2.1 system with the Solo6 serving as the stereo satellites.

Hooking up the Sub6 was simple, and getting a grip on how to tune the whole 2.1 system was far easier than I had expected. By plugging the outputs of my monitoring chain into the L-R inputs on the back of the Sub6, and then taking the L-R outputs up to the Solo6s, I was ready to go. On the rear panel are a number of controls that allowed me to make choices about how I wanted the different components in the system to behave. A hi-pass filter affecting the L-R signals being sent to the satellite speakers can be switched between 75Hz, 100Hz or completely bypassed. A lo-pass filter for the Sub6 can be continuously varied between 50Hz and 150Hz, allowing me to dial in how much of the low frequencies I wanted the Sub6 to handle. There is a volume control for the Sub6, a mute switch, as well as a polarity switch and a phase selector that is continuously variable between zero and 180 degrees. Basically, everything needed to tune the components of the system to each other in a way that best suits the room is on the back of the Sub6. The hi- and low-shelf controls on the Solo6s provide further control. There’s also a jack for a remote bypass switch that brings the sub in and out, while simultaneously defeating and reactivating the hi-pass filter to the Solo6s (or whatever satellite speaker you attach to the Sub6) – a very cool feature for A/B-ing.

Fortunately my room is treated with bass traps, so getting the Sub6 phased to the Solo6s in my listening position was relatively unproblematic (I fear a well-powered 30Hz signal in an untreated room!). After playing with the hi- and lo-pass filters, I settled in on two settings that I liked. One was with the Solo6s in full range mode, and the Sub6 crossed over at 50Hz, allowing the sub to take over where the Solo6s left off (around 40Hz). The other setting I liked was with the Solo6s crossed over at 75Hz and the Sub6’s lo-pass set around 75Hz. I ended up using the latter setting mostly, finding the bass response in my room to feel more powerful with the Sub6 taking more of the low-end duties. Also, with the Solo6s rolled off at 75Hz, the mid range imaging seemed to open up even further, and a gentle extension of the depth of field was noticeable. The hi-end sounded unaffected by whether the Sub6 was in use or not.

Listening to music with the possibility of feeling 30Hz is always a path of discovery. Some records have little to say below 50Hz or so, while others carry low-end signals I never knew were there. The Sub6 delivered the lowest of lows with the same kind of clarity and punch I raved about in the Solo6s. Again, I could hear a lack of distortion, if that’s possible. With bass frequencies that extend deep enough to blow your pant cuffs, there is an inherent lack of definition in some records that blew my mind as well (check out Lenois’ kicks with a subwoofer!). Compared to the Solo6s on their own, the Sub6 left absolutely nothing to the imagination. If there were signals carrying 30Hz, then those were revealed.

Of course, I couldn’t help making comparisons between the Focal 2.1 system I had assembled and the Barefoot MM27s I use at Mavericks Studio. Because the monitoring situations were not the same, I’m not really comparing apples to apples, but I will say that working with the Focal system felt remarkably similar to using the MM27s, but seemed to reach into even deeper frequencies than the Barefoots (who’s low-end response is rated down to 38Hz). The amount of information coming out of both systems across the entire frequency spectrum is remarkable. Still, the Barefoots are a unique experience, since the sub frequencies are delivered on the same physical plane as the rest of the spectrum, whereas the Sub6 feels and sounds more like what it is, a separate subwoofer.

Comparing the Solo6s on their own to the whole Focal 2.1 system forced me to revisit some of my gushing conclusions about the low-end of the Solo6s. Interestingly, the Sub6 made me appreciate the Solo6s low-end even more. Of course, adding 350 watts and nearly twice the speaker diameter added a ton of power and rumble, freeing up the Solo6s to do a slightly better job with the midrange. However, when you consider the price of the Sub6 ($1450 street), we’ve already leapt into another price bracket entirely, and the Solo6s aren’t playing as heroic of a role as they do on their own. Without the Sub6, the Solo6s bring a level of transparent monitoring to a price point where there’s little competition. At $3450 for the whole system, the range of comparable options changes considerably. With that said, the Sub6 was surprisingly easy to add on to the Solo6, and was naturally married to the Solo6s sonically. Anyone doing sound design for film, or working in a genre that requires constant monitoring of the deepest of bass frequencies, would appreciate the clarity, versatility and ease of the Sub6, in use with or without Focal monitors as the satellites.