AudioTechnology App Issue 2

THE GREAT GATSBY: ALL THAT BAZ KORG MS-20 MINI: THE REAL DEAL

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THINKING OUTSIDE THE BOX Getting Analogue from Digital

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Editor Mark Davie mark@audiotechnology.com.au

ED SPACE

Publisher Philip Spencer philip@alchemedia.com.au Editorial Director Christopher Holder chris@audiotechnology.com.au

Digital comes from analogue.

Graphic Designer Leigh Ericksen leigh@alchemedia.com.au

Column: Mark Davie

Art Director Dominic Carey dominic@alchemedia.com.au Advertising Paul Cunningham paul@alchemedia.com.au Accounts Jaedd Asthana jaedd@alchemedia.com.au

Are there any other historical parallels to the recording studio? Where discontinued technologies become so beloved they’re canonised into modern recreations.

There’s the classic car industry. But for the most part, it celebrates vintage without hauling it into the modern age. No one’s going to drop an old 350 Chevy long block into a Nissan GT-R. There’s just no call for it.

Even Rupert Neve, lauded for creating the best analogue gear of the ‘golden’ era, is critical of the past. In a recent interview he lamented consumers chasing after his old designs when his new ones are far better spec’d. Which is why there’s more room in the market for companies dedicated to building ‘Carnhill-equipped’ 1073 clones than Rupert Neve Designs itself.

Even in the movie Sound City — a documentary half about a studio, and half about its centrepiece Neve 8028 analogue console — Neve’s interview was cut to engineering speak, goofy looks from Dave Grohl and overlaid subtitles admitting he had no idea what the hell Neve was on about. The legend reduced to vintage subtext in another example of idolising the man, but not actually listening to him.

In Sound City, digital was the enemy. Old ads about early digital gear were spun into montages like broadsheet Nazi propaganda in a wartime documentary, with straplines reading ‘No Patience Required’. It was an easy target. And two-hour edit renders and marketing schtick didn’t help matters.

The mega-million-unit-selling producer Keith Olsen was portrayed as a weirdo when he jumped ship from Sound City’s analogue haven into an early digital setup. But he still made mega-records, is still in business, and even now uses a hybrid digital and analogue system.

And if we learnt anything from Sound City, it’s that an analogue console and two-inch tape does not Nirvana make.

So, why do we keep coming back to the analogue fountain of youth? And why do the GUIs of latest plug-ins still look like an homage to a 1960s radio station? We’re obsessed with accumulating digital recreations of gear we could never afford, is too rare to find, and probably have never used in the flesh.

And marketers know it.

Here in the office, when we were floating tonguein-cheek ideas of how to illustrate the idea of crafting ‘analogue’ results from digital tools, we came across an ad for Avid’s Heat, proudly labelled Analog Warmth and Color Emulator complete with a glowing ‘warm’ VU meter graphic — couldn’t have done it better ourselves. And that was 2010!

Maybe the mythology helps us make better records, perhaps putting a big tube mic up in front of a singer does help them perform better, likewise a Neve console for an engineer. And maybe that Fairchild plug-in will give you exactly the sound you’re looking for, even if it doesn’t necessarily sound like the real thing.

Better than rote emulation, is deep comprehension.

And that’s what we’re hoping Dax Liniere’s Thinking Outside The Box series will do — take away a dependence on myth, and replace it with a little bit more understanding.

Subscriptions Miriam Mulcahy mim@alchemedia.com.au Regular Contributors Martin Walker Michael Stavrou Paul Tingen Graeme Hague Guy Harrison Greg Walker James Roche Greg Simmons Blair Joscelyne Mark Woods Andrew Bencina Brent Heber

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38

COVER STORY

ProTools 11 Gets Faster-Than-Real-Time-Bounce

18

TABLE OF CONTENTS

All That Baz: Sound For The Most Extravagant Picture Of The Year The Great Gatsby

36

MS-20 Mini: A Legend Reincarnated, Smaller

14

Getting Analogue Results From Digital: Starting On The Right Terms

30

Mix Masterclass: Craig Silvey Dissects Yeah Yeah Yeah’s Art Punk Hit Sacrilege

46

LAST WORD

with Rob Papen

GENERAL NEWS THE FUTURE IS LATER ON Operating in the 2.4GHz frequency band, Audio-Technica’s new System 10 digital wireless is offered as a long term ‘futureproof’ solution for wireless microphones.

System 10 promises reliability and ease of use at an economical price point, starting at A$399 for System10 hand-held, A$499 for a belt-pack with lapel, up to $699 for a system with the Pro92 headworn mic.

Operating in the 2.4GHz range should make the System 10 immune to TV and DTV interference, but it does play in the same sandpit as plenty of other transmitters, including Wi-Fi signals and microwaves.

It has up to eight channels that can be used together without any frequency co-ordination problems or group selection issues. Clear communications are provided by three levels of diversity assurance:

frequency, time, and space. Frequency Diversity sends the signal on two dynamically allocated frequencies for interference-free communication. Time Diversity sends the signal in multiple time slots to maximise immunity to multipath interference. Finally, Space Diversity uses two antennas on each transmitter and receiver to maximise signal integrity.

Multiple system configurations are available, with handheld vocal microphone/ transmitters and UniPak body-pack transmitters designed for use with lapel, headworn and instrument microphones as well as electric guitars.

Technical Audio Group www.tag.com.au

LINE6 MODELS MICS TOO Line6 has made improvements to its XD-V75 digital wireless system by introducing the new Relay V75-SC digital wireless handheld microphone/transmitter with super-cardioid capsule, plus Line6 released updates to the XD-V55HS, Relay G90, Relay G50 and Relay G30 digital wireless systems. Featuring 24-bit audio quality, 10Hz–20kHz frequency response and a wide dynamic range of up to 118dB (A-weighted), the Relay V75-SC promises a performance as close to a wired microphone as is possible at the moment. Line6 is also sticking with offering microphone modelling — a concept that everyone else thought kind of faded away a few years back — with four microphone models available

Audient has launched its new recording interface, the iD22. Features are two-in, six-out + ADAT I/O with 24-bit/96k AD/DA converters, and console-like monitor controller functions in a desktop package. ID22’s Class A mic pres are exactly what’s used in Audient’s standalone units and consoles. You get three programmable function keys (F1-F3) and you can expand iD22 via ADAT. Awave: (03) 9813 1833 or www.awave.com.au

at the press of a button including the Shure Beta 58A, Electro-Voice N/D767, AKG D5 and… the ‘original’ Line 6 super-cardioid microphone (that last one feels like a bit of a cheat). Regardless of whether they sound like the real thing, the instant switch could allow you to choose the mode to match your voice, performance style and venue. Updates for the XD-V55HS digital wireless system include a tan headset option.

The Relay G90, Relay G50 and Relay G30 get a free version 2 firmware update with various performance tweaks.

Australis Music Group www.australismusic.com.au

Based in Canada, Sage Electronics has released the SE-D.I.3 Mighty G DI Box, a palm-size direct active DI box featuring NOS vintage germanium transistors.

The Mighty G is 90mm long and 40mm wide housed in a die-cast enclosure and is designed to plug directly into most instruments. Sage Electronics promise that the Mighty G’s germanium transistors contribute a ‘magical sonic presence’. You can buy the Mighty G direct from Sage Electronics for US$299.

Sage Electronics: www.sageelectronics.com

SSL has unveiled Sigma, a remotecontrolled analogue mix engine.

Designed for the DAW user that seeks the sound of an SSL console while retaining the convenience of working in the box, Sigma is an analogue mix engine in a 2U rack unit that is remote controlled with MIDI via Ethernet using a DAW or an iDevice-compatible software interface. Using proprietary MDAC control technology Sigma’s 100% analogue summing engine can be driven by automation data created within your DAW of choice.

Amber Technology: 1800 251367 or www.ambertech.com.au

The Studer Vista 1 console is now available in a 22-fader version suited to small spaces such as found in OB vans. The Vista 1 still provides a comprehensive solution requiring no additional racks, integral I/O, DSP and surround sound management including up and down-mixing. The Vista 1 is based largely on the Vista 5, so existing Vista users will be familiar with all the functionality of the Vistonics user interface. The Studer Vista 1 also features a redundant PSU and Relink integration with other Studer Vista and OnAir consoles.

ATT Audio Controls: (03) 9379 1511 or www.attaudiocontrols.com

SENNHEISER’S WIRELESS MASTERPIECE BEHRINGER FIRES UP The Behringer factory has come back to life again — things have been almost quiet from the manufacturing behemoth since NAMM, when Behringer pretty much backed up a truck of new gear every day. The latest release is the Firepower FCA610, a half-rack sized six-input/10-output audio interface that offers the handy choice of either Firewire or USB 2.0 connectivity. Also, the Firewire port is six-pin, but the FCA610 comes with an external power supply catering for laptops with four-pin Firewire or USB ports lacking enough juice.

The reality of those six inputs is two ‘Midas-designed’ preamps with combo connectors, two balanced line inputs on the rear and optical/SPDIF

digital ports. Eight balanced TRS line outputs on the rear with the stereo digital out give you the 10 outs. Two separate headphone outs are switchable between the various inputs. Between the switching and possible patching, the FCA610 allows some good flexibility for your headphone or output mixes. You get access to freebie DAW software and 150 effects or instruments from Behringer’s website. If the FCA610 I/O falls short for you, the FCA1616 version has been available for a while, but is a full-width rack-mount and not so portable.

Galactic Music www.galacticmusic.com.au

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EIGHT MORE FOR GLD Allen & Heath has introduced the GLD-112, a larger version of the GLD-80 mixer providing an extra bank with a further eight control strips.

This increases the channel count to 28 fader strips in four layers, each with a motorised fader, channel LCD display that can be named and colour-coded, plus a rotary control for access to gain, pan and aux/FX sends. The GLD-112 keeps the same channel processing control section as the smaller version, complemented by a 214mm-diagonal touch screen. Four additional soft keys are included next to the faders for scene-controls and other userdefined functions. The mixer’s local

I/O is four XLR mic/line inputs, four XLR line outs, four RCA inputs, two RCA outputs, and digital outputs in S/ PDIF and AES3 formats or, of course, the GLD-112 connects to a range of A&H I/O racks. Standard iLive audio I/O option cards for Dante, MADI, Waves and Allen & Heath’s ACE protocols can be fitted, too. The larger GLD is scene-compatible with the existing GLD-80 mixer, using the same audio core. A GLD firmware update version 1.2 has been released, which along with various tweaks includes four new plugin effects.

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SOFTWARE NEWS HAMMER YOUR IPAD WITH THOR One for the Reason fans — and anyone else. Propellerhead has released Thor for iPad, adapting Reason’s flagship synth for the iOS platform with a custom mobile interface and expressive playing features, while promising the same sound. You get an innovative keyboard — meaning something more practical for the touch surface interface (read below) — and the same synthesis engine as its desktop sibling.

It’s the same semi-modular synth with the same set of selectable oscillator and filter types and the same advanced routing of audio and

modulation. Sound same-ish?

That’s the idea. Aside from a regular piano-style keyboard, Thor features a keyboard that can be locked to specific keys and scales. Propellerhead is promoting Thor for iPad as a handy tool for existing Reason users, and any Thor patch created on the iPad can be transferred to a DAW computer, but it is a standalone synth too with Audiobus support, so even if you’re not a seasoned Reason user you can have Thor for $15.99.

Electric Factory www.elfa.com.au

SLATE CHALKS UP THREE NEW COMPRESSORS It’s the last thing we should care about, but you still have to love the graphic detail that Slate Digital puts into its plug-ins. Slate’s latest are three Virtual Buss Compressors (VBC) modelling the nonlinear characteristics of the transformers, tubes, VCAs, amplifiers, phase distortions, harmonic distortions, and timing of real analogue compressors.

The FG-GREY promises a model of the SSL 4000 series bus compressor along with a ‘very small mod’ — we’ll allow that. The FG-RED is a compressor based on the classic Focusrite RED compressor that has been a favourite of mix legend Chris Lord Alge. The third compressor,

Arturia is launching its KeyLab range of professional-grade MIDI keyboard controllers with all-new accompanying Analog Lab synthesiser software. As you might guess, the KeyLab 25 features 25 keys with velocity sensitivity. Larger KeyLab 49 and KeyLab 61 models are available with aftertouch and an additional 16 backlit pressure-sensitive pads. KeyLab controllers come complete with a new version of Arturia’s Analog Lab synthesiser software solution.

CMI Music & Audio: (03) 9315 2244 or www.cmi.com.au

the FG-MU is a plug-in designed to emulate not just a compressor, but the classic tube circuitry behind units such as the Fairchild 670 and Manley Vari-Mu and as such offers tonal colours created by that circuitry before you even begin to start compressing. While all three compressors are available as separate plug-ins in your DAW, you can also use them in Slate’s Virtual Buss Compressors Rack plug-in, in which they can be chained, reordered via drag-and-drop of the rack handles, and soloed independently. All three compressors can be had for US$199.

Slate Digital www.slatedigital.com

It’s taken a while and there’s even an 8.02 update for the Mac DP8 in the meantime, but Digital Performer 8 for Windows is finally here with 64-bit Native operation.

MOTU say it’s ‘nearly identical’ to the Mac version. Features in DP8 are 17 new plug-ins including two new classic guitar amp models, the Subkick kick drum enhancer and the Springamabob vintage spring reverb processor (gotta love that name). It’ll be interesting to see how much a Windows version raises the profile of Digital Performer in the fiercely-contested DAW market.

Major Music Wholesale: (02) 9525 2088 or www.majormusic.com.au

Notable new features in Image-Line’s FL Studio 11 include a Performance mode and multi-touch support for recognised Microsoft gesture functions.

The Playlist has been beefed up to 199 tracks, increased from 99 and overall the GUI has benefited from a host of workflow tweaks. Three new plug-ins in FL Studio 11 are the BassDrum, Groove Machine Synth (GMS) and Effector, while FL Flowstone now uses Ruby high level programming language. Image-Line reckon this means that connected to the right robot, FL Studio is the first DAW in the world that can now make you a coffee. Sounds like the folks at ImageLine need to cut back on the caffeine a little, more than anything.

Major Music Wholesale: (02) 9525 2088 or www.majormusic.com.au

Brief News Version: DTS (originally Digital Theater Systems) and Fairlight have collaboratively announced the release of Fairlight’s new 3DAW, a 3D audio production platform with native MDA mixing and format support.

The first of its kind, Fairlight’s 3DAW enables sound designers to truly mix object-based audio in unrestricted 3D space, monitor on any configuration, and output in DTS’ proposed future specification— MDA. The new Fairlight 3DAW is available as a turnkey solution based on Fairlight’s Crystal Core Media processor and software.

Fairlight: (02) 9975 1777 or www.fairlight.com.au

LIVE FROM YOUR IPAD Apparently it’s been a while coming, but the Live iPad controller from Liine and ST8 is finally here. Built on what Liine learned from making Mu, Griid, Kapture Pad and Lemur, LiveControl 2 promises to give you every single feature you need to jam, produce and perform with Ableton Live. The interface is divided across four pages: Launch, Modulate, Play and Sequencer.

Each page is deep and feature-rich, while the layout is clear and intuitive.

LiveControl 2 is completely free for all Lemur users. Lemur? Lemur is Liine’s professional iOS controller app with a

decent price tag (in the scheme of iPad apps). Any software or hardware that receives MIDI or OSC can be controlled

by Lemur. Control DJ software, live electronic music performance software, studio production software (DAWs), VJ software, visual synthesis software, stage lighting and more… you name it, Lemur is a powerful app and may come in handy for a bunch of other things you’ll want to do.

Regardless, it’s the price you’ll pay for the free LiveControl2 app. Cost: AU$49.99 for Lemur, LiveControl 2 is free.

Lemur www.jazzmutant.com

SOME FIFTIES FLAIR It probably won’t be your everyday, go-to drum sound, but for drum sampler aficionados Native Instruments has added the Abbey Road 50s Drummer to its catalogue of Abbey Road Drummer expansion packs. After the previous Vintage Drummer release, this pack pretty much wraps up the last of the decades. These collections are always recorded at Abbey Road with the attention to detail you’d expect — in this case extra absorbers were used to mimic the tighter recording environments of the 1950s. The drums “feature a rough, cutting sound, with

massive, resounding snares and mellow cymbals.” The two kits recorded were an early-50s Gretsch Cadillac Green Nitron and a late-50s WFL (William F Ludwig) kit. Both have a choice of three interchangeable snares. You get the usual Abbey Road Drummer interface with a full mixer and a library of grooves played in the style of the era plus some modern ones as well. In all, you end up with 20GB of samples. The sampler needs to be run inside the free Kontakt 5 player or the full version of Kontakt.

CMI www.cmi.com.au

PART 1:

START ON THE RIGHT TERMS In this first instalment we bust a few analogue vs digital myths on our way to baking a digital cake that tastes pure analogue.

Tutorial: Dax Liniere

For a long time, there was a clear and undeniable winner of the analogue vs digital, OTB (outside-the-box) vs ITB (in-thebox) debate. Digital technology just could not match the euphonics (those tonal harmonics we find so pleasing to the ear) we’d grown used to in the analogue domain.

But as the understanding of digital signal processing has advanced and processing power became faster and more affordable, ITB has found firm and equal footing with what was once only achievable in the analogue domain.

A FIRE WITHIN

This series of articles is not intended to throw fuel on the fire, but help you blaze your own trail.

While the rest are busy arguing, we can work to better ourselves at our craft and get out-of-thebox results from within it.

To really understand how to get a rich, full sound in-the-box, we need to know what’s really happening to our signals as they pass through analogue consoles and outboard gear.

You often hear people say that songs mixed on an analogue console have more ‘depth’, ‘width’

and ‘punch’ than ITB mixes. As a scienceminded person who understands electronics, I’ve always found some of these descriptions to be a little dubious.

WIDTH — A LEAKY DEFINITION

DEPTH — DOWN TO THE FLOOR

Analogue equipment has an affliction called crosstalk, which is where signal from one signal path ‘leaks’ into adjacent signal paths. This can occur when one or more tracks on a printed circuit board are in close proximity. In a DAW or

Analogue consoles, as much as we love them, are imperfect. They have background noise, present as hiss, and although a professional console will have an acceptably low noise-floor, it still has more noise than well-designed digital audio workstation (DAW) software.

The term ‘depth’ describes how far into the soundfield you can hear, or the contrast between the closest and farthest sounds. The impression of distance or depth is caused by the psychoacoustic properties of volume and delay, hence reverb. As the tail of a reverb decays, it becomes exponentially quieter until, at some point, it drops below the noise floor, becoming masked.

It’s true that the human brain is capable of discerning sounds below a constant, steady-state noise floor, but it also stands to reason that we could hear ‘deeper’ into the soundfield with less noise present.

‘Width’ is even easier to define and can be used to explain away another one of the myths surrounding analogue consoles.

digital mixer, crosstalk cannot naturally occur since each ‘signal path’ is a separate stream of data.

In a stereo mixing scenario, a mono sound that’s panned centre is obviously not coming out of the centre, since there is no centre speaker. It’s created in what we call the phantom centre.

This is simply the psycho-acoustic phenomenon where an identical sound of the same volume and phase arrives at both ears at the same time.

This gives the impression that it originates from directly in front of us.

When you pan a sound hard left, none of that signal is routed to the right channel and viceversa. If you adjust a pan control away from

hard left, you are directing some of that sound to the right channel. The closer to centre pan, the less volume difference there is between left and right channels, until you reach the centre position where both channels receive the same signal and it appears to originate from between the stereo speakers.

If you have a pair of non-identical sounds, panning them less than hard left and right will result in the soundfield becoming narrower.

Crosstalk obviously affects width, therefore claiming a mix done on an analogue console intrinsically has more ‘width’ than an ITB mix is complete nonsense.

PUNCH — CAPACITY FOR ATTACK

While the terms ‘depth’ and ‘width’ are either being used incorrectly, or relate more closely to the skill of the person who can afford to use a large-format console, we still have to address the term ‘punch’. A ‘punchy’ sound can be described as having a strong attack (without being too sharp or biting) and where the attack of the signal is quite even, despite fluctuations in input signal.

There are many parts of an analogue console where distortions and colourations are induced on the signals passing through them. The most significant are transformers and other reactive components such as capacitors.

A transformer uses two coils of wire — a ‘primary’ and a ‘secondary’ — wound in close proximity around a metal core. There is no electrical contact between the individual coils or the metal core — the signal is induced into the secondary coil by the primary coil via electromagnetism. When the coils are driven too hard, i.e. by a large signal, the transformer will saturate or ‘soft-clip’ and introduce harmonic distortion to the signal. In the right quantities, this can be quite pleasing and musical, but as with anything, it can be easily overdone.

Being a magnetic device, the sound of the saturation is somewhat like the sound of analogue tape when it is pushed too hard.

Different types of transformers have different saturation characteristics, mostly depending on the type of metal used in the core (to my ear, the cleanest is nickel, followed by iron, with steel core being the most coloured).

By their nature, low frequencies have more energy than high frequencies. It’s the kind of energy found in a kick drum signal that can saturate a transformer, causing excitation of low frequencies which produces easily audible upper harmonics.

Capacitors are often used in consoles to ‘decouple’ one circuit from another (eg. the mic preamp from the EQ). In many circuit designs the audio output signal contains a portion of the DC power supply voltage and it is not desirable to pass that on to the next circuit. Capacitors are used to make that bridge, passing the audio signal voltage but blocking the power supply voltage. In other words, decoupling the DC component of the signal or removing DC bias.

Dax Liniere is a producer, engineer and owner of Puzzle Factory. A Winston Churchill Fellow, he recently attended Mix With The Masters seminars with Brauer and Chiccarelli, and was in the studio with Alan Moulder while he mixed the latest Foals album.

“

Although a professional console will have an acceptably low noise-floor, it still has more noise than well-designed DAW software

”

Capacitors are reactive components, meaning their behaviour changes with frequency. They are capable of altering a signal’s phase and frequency response, and can therefore impart their own subtle tonal colouration on the signal. Some analogue audio circuits use many decoupling capacitors in each channel strip, collectively contributing significantly to the overall sound of the console. (Transformers are also reactive devices and can alter a signal’s phase and frequency response, in addition to introducing saturation.)

The important thing to remember is that every time your signal passes through another circuit stage, you introduce another layer of distortion and/or colouration.

It’s this short-duration saturation of transients that increases the apparent ‘punch’ of a sound.

Since this soft clipping is due to the input level reaching a finite ceiling, the resulting harmonics are produced at a consistent level.

IN SERIES There’s plenty more to come in this series.

I’ll cover how to control your bottom end, explain many different compressor, EQ, reverb and delay options, guitar amplifier modelling, feature interviews with top plugin developers, plus show you how to get the thickness and cohesiveness we attribute to a good analogue mix.

BANDWIDTH: FLAT TO THE LIMIT

Another difference between OTB and ITB is bandwidth. While console manufacturers and DAW developers all strive to give their products the widest and flattest frequency response, in reality, the limitations of physics come into play in both worlds.

Any digitally sampled signal has an upper limit on the frequencies that can be captured and reproduced. Harry Nyquist, in his sampling theorem, states this is half the sample rate, which for people working at 44.1k, gives a limit of just over 22kHz. Analogue consoles, on the other hand, extend to at least 30kHz. Rupert Neve believes consoles should be ‘clean’ up to 100kHz. Whether those ‘inaudible’ high frequencies are significant is an entirely different argument, but the only way to match this ITB is to run your system at greater than a 192k sampling rate.

On the other end of the spectrum, a properlydesigned DAW has a frequency range that extends flat down to 0Hz. Compared to even the highest spec’d analogue consoles which roll off around 15-20Hz, there’s a fair bit more going on down there, not to mention the phase shift caused by that roll-off.

Though most DAW developers have accomplished what console designers set out to achieve, is that actually useful? Unless you’re using your DAW for scientific purposes, the answer is a resounding no.

AT THE CROSSROADS

Of course, sonic differences aren’t the only thing setting OTB and ITB apart: ergonomics and workflow play a large part in the experience of mixing a song.

A hardware console is undoubtedly a more physical, more tactile way to mix. Even a control surface can’t offer the same level of physicality due to the reduced number of controls. I quite enjoy balancing a mix spread across console faders; it’s a great way to experiment with different combinations far quicker than is possible with a mouse alone.

LET’S BE CLEAR, IT’S ABOUT DISTORTION It’s all well and good to talk about all this theory, but what steps can we take to get the best of OTB while working ITB? In the next issue, I’ll begin uncovering saturators in detail, but for now, I invite you to download the latest free plug-in from Klanghelm, called IVGI. Pop it on your master bus (my preference is directly after the bus compressor) and start with the settings shown in the screenshot. Have a play and a good listen to what it does to your mix and next issue I’ll go into more detail, covering several other plug-in options, plus the allimportant when to use saturators.

TIP: Remember to set IVGI’s output level to achieve unity gain when you toggle bypass, otherwise you won’t get a fair comparison of what it’s doing.

But I’ve observed something interesting. For most ‘A-list’ engineers if the balance of a mix doesn’t feel right… swish, all the faders go back down and they’ll try a different tack.

But when mixing ITB, people seem less likely to start over from scratch, which is odd, because it’s even easier to achieve in a DAW. Just save a copy and try something completely different, if you’re not happy with the road you ventured down, a few clicks and you’re back at the crossroads.

Another difference from ‘the old ways’ is that now we have a computer screen to steal our attention. Whereas before, there were no visual cues to pre-empt, and less distractions from our primary task of listening. Thankfully, there are solutions. Basically, what we want to do is blank our computer monitors quickly and easily when we’re doing our critical listening [which some would argue is all the time — Ed]. On a PC, this

is easily achieved by setting your screensaver to a blank screen. Then use a free program like ShutdownTray to trigger the screensaver with a quick keyboard shortcut. Mac users have it even easier with Hot Corners. Just pick your corner and set it to ‘Put Display To Sleep’.

THE CHOICE IS YOURS

Working in the analogue domain imposes other limitations, but often, those limitations can be used to our advantage. OTB, the number of mixable tracks is limited by the available console inputs and D/A converter channels, and the amount of outboard processors is limited by budget and available space. ITB, our track and processor count is only limited by the computer’s available system resources. Buy a plug-in and you can use as many instances as your system will bear.

When you’re working on a production ITB, don’t leave all of your decisions until the mix phase. While it seems logical that no-one would want to paint themselves into a corner, leaving options open can be an enabler of indecisiveness.

Limitations can challenge you and make you grow. They can force you to think in a different way in order to solve a problem to reach your end goal. Quite simply, practice the act of committing to decisions early in the production process. Sure you might make some mistakes, but without mistakes, how can we grow?

ANALOGUE TIME-SOAK

As mastering engineer Sean Diggins says, “We are in a time-based industry and analogue is a time-soak.”

OTB mixdowns are restricted to realtime; stems are more difficult to produce; and recalls take time and leave room for error. The sound of hardware can change due to heat, power and atmospheric variations, plus deterioration through ageing. While reliability issues can be mostly mitigated through meticulous maintenance, this requires money, time and expertise. Every mixdown, revision, stem and repair takes time. Every extra ‘instance’ of outboard gear costs money, compared to a plugin which is bought once and inserted many times.

To be fair, plug-ins can also break compatibility with system upgrades or lose support completely (as in the case of the fantastic Voxengo Marquis compressor). You also never have to worry about copy protection with outboard gear.

Are the perks of analogue mixing worth the drawbacks, or vice versa? Ultimately, that’s for you to decide. To me, there is a clear-cut winner, and that is ITB. It has everything going for it — instant recall, full automation, ‘unlimited’ processors (restricted only by your CPU power), no noise and minimal maintenance, editing capabilities that open up new sonic and production possibilities, faster-than-realtime bounces and a great sound that can be relied upon to give the same results every time. But remember, to get the most in the box, we must think outside it.

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Story: Christopher Holder & Abigail Sie

The budget is huge, the stakes are high and the pressures are enormous.

AT goes behind the scenes to understand what it takes to work on a Baz Luhrmann epic.

No one’s wingeing. Especially not after you actually see the film.

Love it or hate it (and a Baz Luhrmann film does tend to polarise) The Great Gatsby is a sumptuous production where music and, indeed, sound commands a very prominent position. But working on a Baz film is not just a job, it’s an ordeal. And I mean ‘ordeal’ in the medieval sense of the word: where key personnel are driven to the very edge of their capacities and endurance, then on… right off the scale to the point of physical and mental collapse.

Like I said, no one’s wingeing. Most of Baz’s team has worked with the director before and know exactly what they’re signing up for. And none of this is some kinda trade secret, it’s common knowledge and on the public record just how challenging it is to work with Baz. When you hear professionals talk about Baz, he’s described as a force of nature. A disarmingly charming raconteur. A visionary. An iconoclast.

A reactionary even. Naturally, what’s implied very often is the Yin to the Baz Yang: a side that’s as often infuriating as it is inspiring, a dimension that can simply make you sit in a dark corner and weep as much as it can elevate you to places you didn’t know you could go.

In short, working on a Baz film will leave you utterly spent.

BLOCKBUSTER BUILDUP

AT caught up with a number of the key audio personnel in the weeks after The Great Gatsby’s completion. For these guys there was still some of the demeanour of a hostage victim fronting the media for the first time after a year in solitary confinement.

And it has been about a year. Last March Tim Ryan, Supervising Music Editor, fired up the Gatsby machine at Trackdown Studios, Sydney, where he’s one of the directors/owners. Meanwhile, Sound Designer and Re-recording Mixer, Wayne Pashley, was collecting sounds and working out of his Big Bang Studio in Sydney. But it wasn’t until the beginning of this year that the Gatsby machine really kicked into overdrive. Wayne, Tim and fellow mixer Phil Heywood de-camped to Stage One, at Deluxe in Sydney.

During the ensuing three months the final mix of the key sound components — dialogue, atmos, FX, and music — came together. And here’s where you had to grit your teeth and keep yourself ‘nice’ because the final mix was never final.

What’s hard to describe is how often the goal posts moved. In fact, more than that, how often the goal posts were replaced, or turned upside down, painted mauve, chopped into random toothpicks, and glued back together to form the statue of David. The movie was edited endlessly almost until the moment it was delivered.

FLAPPERS TO RAPPERS

Much has been made of how the music is like another character intricately woven into the film. Leading the charge was Jay-Z, taking on the role of Executive Producer and ensuring the sound track would have the necessary street cred. He, Baz and his long-time lieutenant Anton Monsted, worked to harness the energy and riskiness of modern day artists to evoke the danger and excitement of the fresh jazz sounds of the Gatsby era. Wayne Pashley elaborates: Wayne Pashley: Baz made an interesting call with music. He put himself in the position of Fitzgerald, back in 1922, when jazz was new. Louis Armstrong was just hitting his stride and all that big band material was very new. So with that same concept, he took what Gatsby would be experiencing today in modern music, and put it in this film.

The great thing he did, though, was not just throw in a Jay-Z rap. He combined it with extraordinary material from the Bryan Ferry Jazz Orchestra. And now you have tunes of that era intermixed into Jay Z, Will.i.Am, Fergie, The XX.

It’s a mash of styles that actually does relate to what was probably happening in 1922 in the way Fitzgerald would have seen it; that sort of new age of music and the excitement of it. It’s a great strength of the film.

AT: Was it difficult to pull together something that resembled a common DNA throughout the underscore and the featured songs?

Tim Ryan: It was a big challenge. Often the source cues were augmented by newly recorded jazz overlays, giving them that common DNA you’re talking about. Craig Armstrong’s score you could say lived in a fairly traditional place, as underscore in the movie, and would be used to expose some of the main themes — the Gatsby theme, the Dark Gatsby theme — they could anchor the whole sound track. Then the source music could be the breakouts from that.

But it was those jazz underpinnings that helped to glue it together. And when you marry that to the picture it all helps to make sense.

WP: One of the biggest challenges was the use of voiceover against that sort of music. It’s Nick Carraway [the narrator] telling the story and no one is going to thank you if the voiceover is not heard.

And with so much voiceover it’s sometimes quite a compromise between where you feel you want to push the music and let the voice tell the story in a natural and smooth way. I hope we’ve succeeded there.

AT: Did the music arrive in such a way that it was easy to control in the mix?

WP: It depended on the artist delivering it and their understanding of the requirements of film, in terms of stereo versus 5.1, things like that.

That was probably the biggest deal for us, to make them understand that we needed the mono track so we could then segue into 5.1 on the vocals, for example. And Tim mentions the score [written by Craig Armstrong who worked on Moulin Rouge with Baz], which is absolutely sensational and really acted as subtext. Plus, it was a joy to mix because the themes Craig wrote are so rich, yet had so much air in them that they never stepped on the dialogue.

AT: Music really is the star here.

TR: It’s wall to wall music. In fact, I can only think of one scene in the entire film without music.

WP: Yes, and there’s an interesting story there as well. It’s an eightminute scene and the climax of the film — a showdown of the protagonists — and Baz always maintained there would be no music.

In the book this scene is also set on the hottest day of the year. Each character had a fan blowing on them; porters were bringing in buckets of ice. That was probably the most challenging thing, to get the heat happening and, when it comes to sound, it’s pretty hard to get heat happening without insects.

Basically, New York City and the fans in the room were used like score.

We started by using the city and all its construction and making it musical: horns, the construction of the Empire State Building, distant hammering and sirens. Then we recorded all of the 1920s fans on each character and used the different tonal aspects of each, even down to rattling of the blades and things like that, to increase the tension. Plus ADR, breathing, all those sorts of things, it all came together in the mix and then we would play it back and work out where the moments of tension needed to be put and where we should move away.

The performance of the cast is extraordinary. So I hope we’ve backed them up.

Baz Luhrmann attends a Florence and the Machine session.

TECHNICALLY

AT: I’m guessing in those last three months of the final mix the number of audio channels to deal with was astronomical?

TR: Technically, it was a lot to wrangle. As far as the music goes, each orchestral cue would typically comprise 15 stems, 5.1 wide (90 tracks).

Then there would be overlaps [one cue running into another] with the same number of tracks again (180 tracks). Then we would have the breakout tracks as well that could easily be 100 tracks wide, what with multiple vocals, jazz overlays… the trackcount very easily racked up.

I recall for Spool 2 we hit the 512-voice wall a couple of times and had to strip it back.

For the final mix we would have our own music rig on the mix stage. I would be there with my pre-mixed music mixes, and I’d send stems to Wayne and Phil via 48 outputs from my ProTools I/O. That might sound like a lot to have active at any one time. But we needed to keep all our options open, even so late in the process. The demands on Baz were such that it was a struggle to get enough review time with him and sign-off on the finals. Combine that with Baz’s penchant for constantly reinventing material right up to the 11th hour and you really did need to remain as flexible as you possibly could.

WP: It was a lot to deal with but I have to give credit to the crew and how efficient we became. We could turn around a 20-minute spool in a day. Besides the conform [realigning/adjusting the audio to match the newly edited visuals], suddenly we might spot a steam train passing through an environment that wasn’t there before. So the steam train would get tracklayed and we’d hit the mixing stage (StageOne at Deluxe) by 5:30 that afternoon, we’d conform the Neve and we would mix it in live, no predub first, and we’d be done by 8:00. So that’s how fast we were able to pull it together. That went on for three months.

AT: How did you tool up to deal with the onslaught, Tim? TR: When the workload was ramping up in 2012 and Craig Beckett joined me at Trackdown as a music editor, I realised there was no way — with the amount of material coming in — that we were going to be able to maintain our own local versions and keep them in sync. So we installed a Synology NAS (server) system.

Also around that time we upgraded our ’Tools rigs to HDX2 on fullyloaded Mac Pros. I could see that there was no way we were going to be able to cover this with 256 voices. One thing that made life faster and easier to respond to changes, was to have the tempo maps from all the orchestral recordings — any tempo maps we could get our hands on.

Work in tempo maps always made things a whole lot faster.

The other thing to note was the project was entirely in 48k. I don’t think we could have done this film in 96k. It would have been too wide.

THE ART & SCIENCE

AT: Just to go back a step. Wayne, how did you approach the overall sound design of this film?

WP: The directive from Baz was quite simple but very potent. It was: Protect Fitzgerald.

Basically there were two authentic approaches as described in the book. One was the truth of New York back in 1922, which was a progressive city, full of life, a city of construction. Buildings were going up faster than any other city in the world. Then there was the Long Island set which were the rich establishment living in palatial escapes.

Dialogue was king, particularly in this case – you can’t afford to miss a word. Tom Buchanan, Daisy Buchanan, Myrtle Wilson, George Wilson – and with their dialogue came the embodiment of those words which is everything they’d wear. The costumes of the women, the jewellery, cigarette lighters, were all recorded as sound effects quite separate to foley which gave them a sense of great focus in terms of the Nick Carraway point of view [Nick being the book’s narrator]. I want to do a spoiler here because Baz has taken the fact that the book is written in first person and done an interesting visual device that allows the audience to experience Nick Carraway‘s subjective point of view of the world. Sound had to support that.

AT: And from your perspective Tim, given the importance of music to the film what’s the role of Music Editor?

Tim Ryan: The edit is almost always picture driven — the music is chopped where the visual cuts are. So my job on receipt of the new cut is to try and restore the musicality of the edit; restore the melodies and beats so it’s musically satisfying.

It’s not always easy and you always have edits where you’re not entirely satisfied. You can do some little tempo changes under the bonnet that as a punter you may not notice, where the ear is distracted by the eye — there are workarounds.

Then there are some of the more satisfying musical edits. Early on in the film there’s the introduction to Gatsby, all set to Gershwin.

That was a matter of making a three-minute Gershwin item out of a 17-minute performance. It was a challenge to make that as musically satisfying as possible.

Conforming the new edit to the music is a manual task. There are autoconforming tools available to dialogue editors that work very well, but for music we can see where the music begins and starts from the guide tracks and we make those edits in between with an ear for what’s most musical and rhythmic.

BRING OUT YOUR 8 CYLINDERS? DAMN STRAIGHT! Wayne Pashley: We recorded about 15 or so vintage vehicles, from Rolls Royces through to a hero vehicle in the film that is called the Duesenberg Straight 8. Now that car, our hero vehicle, is a very important prop in the film. I suppose you could liken it to the DeLorean in Back to the Future. Warner Brothers actually bought two of them — but they were replicas and had V8s in them.

Baz, in his enthusiasm for attention to detail, wanted the original straight 8 engine, which is similar to a Mack truck.

But recordings of that original engine didn’t exist.

There are only two places in the world that had these vehicles.

One was in Indiana and the other is at Jay Leno’s garage, in Burbank. Now Jay has great enthusiasm for vehicles. He’s poured a lot of money into his incredible, museum-like place.

He actually has eight Duesenbergs, all of them restored to spec. He buys vehicles that are run down from some dusty old garage and falling to pieces. Then he gets the blueprints and re-machines the engines back to original form.

This particular one we wanted was also supercharged. So we went to Jay Leno and he agreed to let us record it with a donation to one of his favourite charities. He actually spent the day with us. We ripped the

car from head to toe, hooning around Bob Hope Airport.

So I just think it’s wonderful to have that sort of magic, that sort of degree of authenticity in gathering sounds for the film and the Duesenberg is one of them. The film is probably the only recording of that sort of vehicle in existence.

Sound Designer Fabian Sanjurjo: The main Duesenburg scene sees Gatsby driving through Long Island all the way to New York City. That was a huge undertaking. I tracked up our hero car with Jay’s original Duesenberg sound. That was the main sound for the car. But it still needed enhancing — I mean, Jay’s car is a great car but it doesn’t have that high-performance-sounding edge that Baz was looking for. It needed sweetening. I went through shot by shot and added elements such as the sound of animals roaring, bears and big cats, on the passes.

We also added the sound of a supercharger that we recorded quite separately from the car, and stuck it on top.

I also used recordings that we did of the replica Duesenburg for high intensity passes. All in all it was just a building up of layers on top of the authentic Duesenberg. So what you hear in the film is Jay Leno’s car with all this sweetness added to it as well.

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RE-CONFORMING & CONFORMALIZER

Sound editors would love to work to a final picture edit, but it’s rarely the case, especially now with films being made with so many visual FX shots. The picture edit keeps getting updated, often right up until it goes to final mix and dubbing (exporting) for cinema prints.

Audio ‘re-conforming’ refers to the process of re-fitting the sound to the updated picture edit.

Traditionally sound engineers would have to find the changes by eye and move everything by hand, one laborious step at a time. There might be thousands of edited source

tracks on half a dozen ProTools machines, hundreds of tracks of 7.1 premixes, final mix stems.

Conformalizer (a program by Justin Webster) is an audio post production tool which automates both the process of comparing the picture cuts and the process of ‘re-conforming’ the audio to the new version of the reel.

Using EDLs or ‘edit decision lists’ supplied by the picture editor it can accurately find every little change, including CGI tweaks, thereby reducing human error and allowing the editor more time to focus on creative stuff.

Wayne Pashley, Fabian Sanjurjo and co. employed a variety of mics including the Sennheiser 816 and 416, Electro-Voice RE20, and Countryman radio mics. Normally, elements such as the sound of vintage fans, a supercharger, or a old-school typewriter would be the domain of the Sound FX Editor, but as Sound Designer Wayne Pashley wanted hands-on control.

Photos: Jenny Ward

3D WORLD

AT: Wayne, how does the 3D finish affect the way you tracklay and mix?

WP: No matter where you care to look in that 3D world, you’ll be able to find a little sound to match. And that level of visual detail impacted on our mix. It’s such a busy screen, in terms of Catherine Martin’s amazing production design, that the eye could easily wander over here when in fact you want the audience to look over there. So we used foley, particularly foley ‘star’ sounds to help focus the audience. And because of the richness of detail in the 3D rendition, we found we could really push the surrounds so much more, actually lean on them, which helps envelop the audience into their world.

The environments, the atmospheres; quite a lot of times we were pushing the strings of the music to the back. Also a lot of hard FX and hit-points in the music were raised, to kind of push you into the world.

The production design in the 3D is extraordinary.

EXTRA MILE

AT: So why go through the agony? Why endanger your health (both physical and mental) to be involved in a Baz Luhrmann film?

TR: The simple answer is: Baz can pull together a big budget production and how many are there of those in Australia? There’s him, George Miller, Peter Weir perhaps… they don’t come along that often.

As a result, people will put in the extra work for the opportunity. It has to be said that Baz’s ‘shoot for the stars’ approach does inspire people.

WP: I’d support that. The inspiration from Baz as a leader is just extraordinary. We were all willing to go the extra mile because we knew we were working on something very special.

AT: Still, it takes a special kind of individual to thrive in that sort of environment.

WP: Sure, and we had a wonderful mixing crew with Steve Maslow and Phil Heywood. Phil was on effects and Steve on dialogue and music. I’ve worked with Phil Heywood for over 15 years. And, of course, I have worked with Steve before and knew he was one of the coolest guys on the planet, so I knew that he would work for Baz. Steve’s prowess is in dialogue mixing and particularly EQ – I’ve never seen anyone as good at it. His command of the room in the final mix was tremendous – his ability to maintain a cool head was invaluable. It was a great team and one I hope to work with again because they were so much fun and so easy to communicate with.

AT: Writing about a Baz film is also tinged with sadness, given it reminds AT and our friends that Simon Leadley is no longer with us. [Simon was one of the founders of Trackdown, long-time AT contributor and veteran of a number of Baz films.] Tim, you worked with Simon for more than 25 years what was it like doing Gatsby without him?

TR: Simon always had a great overarching view of the whole production — something sorely missed. I certainly did find myself at times thinking: ‘What would Simon do in this situation?’. I could find the answer there; use his guidance in what I thought he would have done and that certainly helped get me through. At the end of the day I am sure he would have been proud of what we managed to achieve.

AT: It’s only been a few weeks since finishing, but what are your reflections?

TR: It’s a genuine challenge. You learn a lot about yourself and the profession you’re in. The lessons you learn? You have to: always be adaptable, never be precious, make changes quickly on the fly, know your material and try to understand your director.

Thanks to the Australian Screen Sound Guild (ASSG) for its help with the Wayne Pashley interview.

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FEATURE

A little Mathew Watson’s classic EMS Synthi AKS synthesiser unearths a whole generation of Dalek nightmares and Other Places.

Story: Jason Allen

If you’ve heard Pink Floyd’s Dark Side of the Moon, you’ve heard an EMS Synthi AKS. It was used by David Gilmour and producer Alan Parsons to create the classic synth track On The Run. And if you’ve had nightmares about Doctor Who’s Daleks, they’ve probably been soundtracked by an EMS synth.

The London company’s synths powered the BBC Radiophonic Workshop into the ’70s; Brian Eno, Yes and David Bowie were all EMS fans; and the company led the world in the production of affordable synthesisers for gigging musicians.

EMS was the UK’s answer to Moog.

Following on from the successes of EMS’s original design — the affordable VCS3 synth — and the portable Synthi A version, EMS released the Synthi AKS. Positioned as an expanded version of the Synthi A, the AKS included a membrane keyboard and sequencer mounted in the lid of its Spartanite case, hence the ‘KS’.

In 1972, at a time when a modular synth could

be the size and price of a small house, the Synthi AKS sold for around £450 — the equivalent of around £5000 today.

Compared to the processing power available now, the AKS was a relatively simple beast — three oscillators, noise generator, ring modulator, lowpass filter and a spring reverb. It has two control voltage sources, including an envelope generator, accessed through a joystick and a trigger button respectively. Two mic inputs and two line inputs allowed you to use the Synthi as a powerful sound processor, and, due to the difficulty musicians not versed in synthesis had using EMS products as melodic sources, a lot of units ended up not doing much more than that.

THE AKS PALETTE

One of these legendary synths has found its way to Melbourne’s Mathew Watson, who is about to release his second AKS-based album as Other Places. Mathew’s Synthi AKS has an impressive lineage — it has passed through the hands of

“

If you’ve had nightmares about Doctor Who’s Daleks, they’ve probably been soundtracked by an EMS synth

”

famous TB-303 circuit bender Robin Whittle to Mathew’s mentor, electronic music pioneer Ollie Olsen. Mathew was first captivated by the possibilities of the AKS when Ollie temporarily expanded his duo Fading Fires. For a time, the live ensemble became a trio with Ollie building rich soundscapes via his AKS and a loop pedal around Chris Rainier’s treated lap steel, and Mathew on drums and Roland SH-09.

a game of Battleship. It may look complex, but it’s actually quite simple when broken down.

Creating sounds with the AKS is similar to mixing paint on an artist’s palette — from the selection of six or so carefully chosen paints the artist knows they can make thousands of colours.

The AKS is so raw and brutal. It’s stepping back in time to before electronics were refined. The sounds it makes are just so exciting.”

That AKS now has pride of place at Mathew’s studio. “I never asked to buy it, not once. I always saw it as a part of Ollie,” explained Mathew. “A mutual friend mentioned to Ollie that I was really inspired by and in awe of it. Ollie then offered it to me (for a price!) with the instruction that I should never sell it. I feel the synth will always truly belong to Ollie but I am the caretaker.” Its abstract, physical nature took Mathew a lot of experimentation to come to terms with, but this has led to an almost poetic understanding of its character. “When people who aren’t familiar with it look at the AKS, they think it looks like

MELODY MODULE

The history of EMS synths shows a general skewing towards out-there cosmic soundscapes.

Probably because of the brand’s close ties to The BBC Radiophonic Workshop (EMS co-founder Peter Zinovieff played in Unit Delta Plus with Workshop alumni Brian Hodgson and Delia Darbyshire), which used re-rigged test oscillators and the Wobbulator for sound generation before EMS came along. And many of the German Kosmische and Krautrock artists of the ’70s — including Faust and Klaus Schulze — embraced the chaos of the VCS3, Synthi A and Synthi AKS

to execute their psychedelic visions. Composer Tristram Cary’s (also an EMS co-founder) sci-fi soundtracks like Hammer’s 1967 Quatermass and The Pit featured EMS synths, as well as a long run of Doctor Who credits — including those devilish Daleks.

Mathew was smitten by “the mystical, otherworldy nature of what their music was about and in turn, the instruments they made that music on”. But when he added an EMS DK-1 monophonic keyboard controller, the EMS world of sound opened up far beyond those cosmic expositions… into melody. The DK-1 was originally released in 1969 as an interface for the VCS3, and gives the player 37 keys, an extra oscillator and two control voltage outputs.

“Other Places started the moment I got the DK-1,” said Mathew. “The AKS is not generally used as a monosynth. The more I was told it was purely for sound effects and processing, the more determined I became to extract melodic behaviour from it.” Mathew doggedly pursued

ENTER THE MATRIX The true power and flexibility of the Synthi, along with its perceived complexity, come from EMS’s revolutionary 16x16 patch matrix. It effectively condenses the wall-sized patch bays of full-sized modular synths into just a few square centimetres. Connections are made with tiny colour-coded pins, which are actually resistors — white for 2.7kΩ with 5% tolerance, red for 2.7kΩ with 1-2% tolerance and green for attenuation with 68kΩ of resistance. This means you can patch anything to anything else; control a filter with an oscillator, an oscillator with the envelope generator, feed the result into the reverb and feed the reverb back into itself. The results are raw, rich and complex, and, due to the variability in the resistance between patch pins, the temperature of the unit and the crosstalk between circuits, sometimes unreproducible. Over the years, this unpredictability has fuelled the mystique surrounding EMS products, making them one of the most sought-after brands in the retro and vintage market.

his contrarian line, learning to make the AKS produce soaring leads and dirty bass sounds.

“The quality of its sound is so unique, but as soon as I had a means of playing melodic music, it opened everything right up.”

COSMIC CUT ’N’ PASTE

Inspired by, but not limited to the technology of the ’70s, Mathew procured a laptop and a copy of Ableton Live and got to work. “The first album eventuated out of six months of trial and error.

I’d play a line in from the AKS, which would then inspire a drum pattern or rhythm. Sometimes I’d start by setting up something simple in Ableton using a Drum Rack with a Roland TR-808 or 606 emulation and then come up with a textural or melodic idea.” Mathew laid down idea after idea, filling his hard drive with drones, leads, basses and grooves. The narrative of the album began to take shape. “I liked the idea of something futuristic and cosmic. I wanted it to sound like the BBC Radiophonic Workshop with Keith Moon playing drums,” Mathew explained. Luckily for his inner-city Melbourne neighbours, Mathew retired to his hometown of Stawell in western Victoria to work on his drum parts: “My uncle offered me his garage. I’d take all my drums and mics and my laptop and just experiment with different drum ideas.” The drums for the album were then tracked with engineer Nick Treweek (Augie March, Glen Richards) at his Fairfield studio.

Mastered by Byron Scullin at Deluxe Mastering, the self-titled Other Places dropped in November 2011. It garnered strong support from Australian independent radio, a feature interview on Germany’s national broadcaster Deutsch Welle and landed Mathew a support slot for New York math-rockers Battles. The critical acclaim didn’t hurt Mathew’s career as a drummer, either. As the album hit the airwaves, Mathew was helping lead 110 other drummers in the 111BoaDrum astronomical happening conducted by Japanese

experimental legends Boredoms in Byron Bay. In 2012, Mathew performed as part of Yeah Yeah Yeah’s guitarist Nick Zinner’s 41 Strings/IIII project at Sydney Opera House for the Sydney Festival, a piece inspired by Earth Day and Vivaldi’s Four Seasons.

TAKE TWO

Mathew has now gone back into the studio and is completing his second Other Places album. From the album’s inception, Mathew has thoughtfully tackled one of the central questions facing all musicians now working with synthesisers and computers — what is the best and most productive way to create music when you are faced with infinite possibilities? And Mathew is conscientiously applying the lessons learnt from album one.

“The thing I took away from the first album was that I wanted to develop my process,” continues Mathew. “How do I set up my studio so I can sit down, turn everything on and make music?”

A time-honoured strategy composers have found for the problem of too much freedom is to selfimpose limitations. Stravinsky summed up the problem nicely in Poetics of Music: “I experience a sort of terror when, at the moment of setting to work and finding myself before the infinitude of possibilities that present themselves, I have the feeling that everything is permissible to me… My freedom thus consists in my moving about within the narrow frame that I have assigned myself for each one of my undertakings.”

Mathew’s ‘narrow frame’ is a setup with the AKS at the core, Ableton as drum synth and recording medium, a Line 6 loop pedal to facilitate a live performance-based workflow and a Dave Smith Instruments Poly Evolver synth for extra processing and textures. The key is having everything patched to everything else — modular, flexible and ready to perform at a moment’s notice. “I now have a constant setup,” said Mathew. “I always have everything connected. If I’m getting stuck creatively I can just feed a signal into the looper, or send a line out of Ableton into the AKS. If I feed a drum machine from Ableton into the AKS and patch it though a filter while I’m also playing a melodic line, it just explodes. I wanted to get more into an organic way of creating electronic music. I’m responding to what I’m hearing, not a static mix.”

By holding himself to this discipline, the act of writing becomes easier, but most importantly each session nets useful music. “You can noodle for years on end, have a hard drive filled with sounds, and build music by layering those sounds, but the music you make like that sounds a certain way — it’s cut and paste,” said Mathew. “I’ve found that the best creative moments are born out of mistakes and fleeting moments of weirdness.” www.otherplaces.co www.soundcloud.com/other-places www.otherplaces.bandcamp.com

Mathew Watson runs AudioTechnology through the EMS Synthi AKS and what it can do. http://youtu.be/wc8r_3OC1qk

JULY 23-25

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ALIA

TUTORIAL

Born in the California Bay Area, Craig Silvey cut his engineering teeth working in local studios and doing an internship with George Massenburg. Silvey went on to become chief engineer at George Lucas’ Skywalker Sound scoring studio from 1992-95, during which time he worked with a number of legendary producers, among them Massenburg, as well as Phil Ramone and Walter Afanasieff. According to Silvey, this meant that a lot of his education came from “guys who had their training in the ’60s and ’70s.” He became a self-declared “obsessive gear nut” and “analogue freak,” and after leaving Skywalker, put the extensive collection of gear he had gathered to good use in a San Francisco studio he called Toast.

He’s been based in London since 2002 and calls himself, “some kind of weird hybrid now: half-American, half-British.”

During 2009-12, Silvey worked at a studio in London called The Garden, but he recently moved to a studio in West London, also named Toast.

It’s filled to the brim with antique and vintage gear that he has collected over two decades. Pride of place goes to his 40-input Neve 8026, which is loaded with 1084 and 1076 modules and has 40 inputs and a 24-channel monitor section.

There’s also a Neve BCM10 sidecar, and a wealth of outboard, including EQs by Helios and John Hardy, compressors such as the Thermionic Culture Phoenix, Universal Audio 1176 and Urei LA4A, mic preamps by Telefunken (V76, V72), Summit and Tube-Tech, a whole posse of vintage mics, and his beloved Mastering Labs monitors, the legendary ML10s. Recent additions include a couple of Pye compressors, several Neve 32254 modules, Cartec EQP1A units, and the Dramastic Audio Obsidian TX10 bus compressor.

Yeah Yeah Yeahs Sacrilege

Some would argue Craig Silvey’s attachment to analogue is motivated by nostalgia rather common sense, but he continues to be one of today’s most successful, and fashionable, mixers

as is demonstrated by recent credits that include Goldfrapp, Florence + the Machine, Anna Calvi, Frightened Rabbit and Yeah Yeah Yeahs. Going further back in time he’s also worked with REM, Santana, Ray Davies, Nine Inch Nails, Linda Ronstadt, Aaron Neville, Pearl Jam, The Coral, Portishead, The Magic Numbers, and Arcade Fire. Silvey’s working methods clearly remain relevant, well illustrated by his mix, while still at The Garden, of most of the recent Yeah Yeah Yeahs album, Mosquito — including the album’s opening track and first single Sacrilege.

All but one of the 11 songs on the New York art punk trio’s fourth album were co-produced by Nick Launay, and David Andrew Sitek of the band TV On The Radio. Both have rather opposing backgrounds and methods. The former has worked with Public Image Ltd, Killing Joke, Kate Bush, David Byrne, INXS, Eric Clapton, Lou Reed, and Arcade Fire, and the latter with Liars, Foals, and Beady Eye. “Probably the most difficult thing for me in mixing the Yeah Yeah Yeahs was trying to balance Launay’s and Sitek’s very different approaches and make them sound consistent,” notes Silvey. “The tension between their approaches is part of what makes that record really good. Sacrilege is a good example.

The bare bones, gritty, lo-fi live element that starts the song comes from Launay, and it then builds up with a disco bass, jazzy guitar and some delayed drums that come in halfway, which are more Sitek’s direction. His stuff is more polished and wider in the stereo field. And then two thirds into the song there’s a totally unexpected introduction of a choir, which turns it into a pure gospel tune.

During the recordings Sitek and Launay were each in a different studio, doing their thing at the same time. They were bouncing stuff back and forth and the band decided what they liked about what the producers were doing. I think I was brought in as an extra opinion, to draw it all together.”

1. A MOMENT IN TIME SAVES NINE

“When you mix in the box, you can do recalls of several different songs on any given day, weeks after you mixed them.

But my equipment and my brain don’t work that way. And in my opinion, music suffers from that approach. Music is a moment in time, and the same regard should be kept while mixing. When you mix a song for the first time, you should focus on how it makes you feel rather than how to achieve perfection. If you enjoy it, and it makes you dance around the room, it’s great. But when you hear it two weeks later and start wondering whether the volume of the hi-hat is correct or not, you have to stop yourself and wonder whether it’s relevant. It may not be. Sadly, perfectionism is becoming the norm now, so it was really refreshing for me to do something like Arctic Monkeys’ Suck It And See (2011), which was recorded entirely to analogue tape, and I mixed from 24-track tape. There was no ProTools in the room and without a screen to look at we had to use our ears again, and learn the music to be able ride the vocals effectively!”

2. THE BUILD UP

“I mix in waves. The first time I speed through a mix, trying to get a sense of what’s there and a direction for the song. I then will start on the drums, but won’t spend a lot of time on them. I may get them to 50% of where I want them, and then quickly move on to the bass, and then the guitars, keyboards, vocal, and so on. I’ll then listen to the whole and do individual tweaks. I’ll then go back to the drums in solo, the bass in solo, and so on, and I may repeat that process two or three times.

From that point onwards I’ll listen only to the complete mix. In the case of Sacrilege, I tried to get the vocal sound right from the get-go. I built the track up gradually, getting everything up to 30%, then 60%, and then to 90% in terms of effects and quality. You can’t really decide what the drum sound is going to be until you know what all the other things sound like.

Plus a track like Sacrilege really has to have the right dynamic shape from beginning to end, because it changes so much. So there was a lot of listening to it right from the beginning with everything in, and making sure it grew in the right way. I’d work on something, and after pressing stop, I’d go right back to the beginning again.”

3. DON’T DISCOUNT PLUG-INS

“I obviously prefer to use outboard, but there are two plug-ins that I couldn’t live without. If there’s a scene change on a track, like a kick drum that changes sound, I may do that in the box, and if I have seven kick drums or so, I’ll do a global kick drum on the desk, and may adjust individual kick drums with a plug-in. I had a Massey TapeHead tape simulator plug-in on the hi-hat and on several of the percussion tracks in the Sacrilege session. It is super cheap, but actually achieves the closest approximation of what tape does. It really does that thing of smoothing out the harmonics on instruments like tambourine and hi-hat that are difficult to get to sound both loud and pleasant in digital. Plus I had the Massenburg Design Works EQ5 parametric EQ plug-in on the drum loops, used correctively. On the desk the drums were grouped together and sent to a bus compressor, the GML 8900 — used for parallel compression.

I used Neve EQ on the desk, and also a reverb called the Great British Spring, which is very cheap and looks like a drainpipe. I like it a lot. Plus I had the Yamaha SPX90 for early reflections to give some more spread and thickness.”

4. WHEN ALL THE OUTBOARD IN THE WORLD...

“The main bass track is called ‘Nakamichi bass’.

Launay and Sitek had used one of those old, high-end Nakamichi cassette decks, which is six rack-units high.

They didn’t actually run the sound through a cassette, but just overloaded the front end, so the distortion you hear is in the Nakamichi preamps. The producers also used the Nakamichi to heavily distort the lead vocal. I put the bass through a little bit of LA2A compression for some warmth to add to that distortion, and used EQ on the desk. I also at times probably added a little bit of sub harmonic synthesiser from a dbx 120A.

“

Probably the most difficult thing… trying to balance Launay’s and Sitek’s very different approaches and make them sound consistent

”

5. NASTY VOICES

“There are two main vocal tracks, the top one, in brown, called ‘Chico’, was recorded through a Roland Chorus Echo tape delay, and this is the clean vocal. Below that is the ‘Nakamichi’ vocal. The Nakamichi vocal track was only occasionally used, and I had to make sure the clean vocal didn’t sound dramatically different, so in certain places I added the SoundToys Decapitator plug-in to the clean vocal to make sure it had some edge to it . I also used outboard on the vocal, like the Empirical Labs Distressor to help bring her vocal forward, some Cartec EQ to add some presence, and a small room sound from the Ursa Major Space Station for early reflections and to create some three-dimensionality. I also wanted to give the choir a nastier sound to make it fit with the song, so I added more mid-range using desk EQ, and I cut out some of the low end so it would not sound as full as a choir normally would. I then applied the Lexicon Super Prime Time to the choir, to blur it a little bit, and a Decapitator, so it wasn’t as in your face and hi-fi as on a gospel record.”

6. MASTERING TO VINYL

“Mastering to vinyl involves EQ-ing as one would do for digital mastering, and then pressing a 12-inch, 45rpm lacquer for each song, with maximum groove width, so you get the best-sounding vinyl possible. Then you do a single playback and record it back into the computer. It’s really hard to explain what it does, but somehow the low end surrounds you more. It also adds some harmonic distortion at the top that gives a very comforting character

— certainly for my generation, which has grown up with vinyl. Many mixers run things to a tape machine before or after mastering, and I have tried that as well, but in my opinion you lose more than you gain. I’ve used vinyl mastering for several records I’ve mixed recently, including Yeah Yeah Yeahs, The National and Goldfrapp. The former two were mastered by Greg Calbi at Sterling Sound in New York, while the Goldfrapp album was mastered by John Dent at Loud in Cornwall, in England.”

REGULARS

MAC NOTES Is the new Mac Pro’s reliance on Thunderbolt a shock to the system?

Column: Anthony Garvin

Causing quite a buzz amongst professional Apple users, the new Mac Pro announced at the June Worldwide Developers Conference seems to have opinions divided — not least over the svelte gloss-black, undeniably dustbinlooking enclosure.

Apple has taken quite a leap with the reincarnation of its flagship computer, requiring many users to upgrade more than just their computer if they buy in. But is Apple’s approach any surprise? Apple has a history of taking plunges like this — moving to Intel from PowerPC processors in 2005-6 and rather controversially dropping the serial and ADB buses in favour of USB in 1999. Hmm, life without USB nowadays would be unthinkable…

Here are some pros and cons of the new machine from a DAW user’s perspective: THE CONS

In dramatically reducing the size of the computer, Apple has dropped capability for any second, or third, or fourth, internal storage drives. The lack of these extra drive bays in the machine means that external storage will most likely be necessary, upsetting in particular the Mac Pro users who currently travel with an all-in-one tower setup.

For those not travelling with their computer, it is also a pain, as who wants a sexy looking computer surrounded by a plethora of external hard drives, cables and wall wart power supplies?

Further to the internal expansion capabilities — or lack of — PCIe slots have been dropped altogether. Quite painful for ProTools HDX users (and Avid) to speak of one large group.

THE PROS

On paper, the increased processing power of the new Mac Pro is a good step forward. With up to two times faster processing and similar memory bandwidth boosts, the machine should be significantly faster than current Mac Pros.

Also on the topic of performance, the internal storage in the machine has been re-engineered.

Firstly, we have SSD storage, which is already

much faster than HDD storage (albeit more expensive per gigabyte — is that a Con?).

Secondly, this SSD storage now interfaces with the computer directly via the PCIe bus, rather than via SATA, which opens up a much, much wider pipeline for data transfer. I’m imagining that any DAW project, regardless of track count, plug-in or video use, will be able to run off this internal drive at the same time as the OS, without issue — a big no-no in the past.

Despite the lack of PCIe slots, the machine is going to be greatly expandable, due to the provisioning of four USB 3.0 and six new Thunderbolt 2 ports (allowing for six devices to be attached to each port, totalling 36 devices).

MORE ON THUNDERBOLT 2

Following on from my previous rants about Thunderbolt’s potential, I’ve been very happy to see that Apple and Intel have been developing this further. And in some ways, I’m optimistic about PCIe expansion slots being dropped in the Mac Pro, as it should force hardware developers to come on board with Thunderbolt very promptly.

The Thunderbolt 2 spec boosts the capability of the interface significantly — well, at first glance anyway. How it actually works is the total bandwidth across the interface is now one upstream and one downstream channel of 20Gbps each, whereas Thunderbolt 1 was two upstream and two downstream channels of 10Gbps each. So, there is no bandwidth gain overall, but one device can now have up to 20Gbps of bandwidth solely available to it — both to and from the computer.

Let’s put that into perspective. Early tests from Intel have shown that a drive array can transfer data via Thunderbolt 2 at 1100MB per second. That is 1100 megabytes per second! To simplify, you could push the data for about 3000 24-bit/96k audio tracks in real-time across this and not hit the maximum bandwidth (OK, I know it’s not that simple in the real world with audio interfaces and the like, but you get the

idea). Intel also claims that these are early days for the specification and it expects to push that up significantly by the time it is released.

So, coming back to my point about the expandability of the Mac Pro, with the six ports of seemingly limitless bandwidth, and a technical capability for six devices to be attached per port, that is the capability for many, many audio I/O interfaces, DSP boxes, working drives, backup drives, backup-of-backup drives, video interfaces, iPhone chargers, desk lamps, etc.

The only problem here — and perhaps this should go back to a Con, is that we are missing a few pieces of Thunderbolt kit at this point: • ProTools HDX ‘Thunderbolt’ (Avid has only released a HD Native Thunderbolt option to date).

• RME HDSP ‘Thunderbolt’ (for everyone needing large numbers of I/O but not using ’Tools).

• A variety of reasonably-priced Thunderbolt hard drives (like the current variety of Firewire options available).

So, having said all this… I think the success of the new Mac Pro, from the public’s point of view, comes down to the price Apple will sell it for. To this point, it has been tight lipped — hopefully because it is still too early to finalise — but my prediction (and hope) is it will match the price of the current Mac Pro, starting at $3000 for the base model.

REVIEW

KORG MS-20 MINI Who’d have thought Korg would go this far back to the drawing board. Some things need never change.

Review: Brad Watts

MS-20 ARCHITECTURE

NEED TO KNOW

The Korg MS-20 is an analogue two-oscillator monophonic lead and bass synth with hard-wired and patchable connections. This ‘semi-modular’ approach means the synth will function without any patching (although you’ll need to do some work if you want the mod wheel to operate) but you can over-ride the hard-wired architecture via the patchbay. In addition to two analogue oscillators, the MS20 features two resonant filters, two VCAs, sample and hold, a noise generator, and that assignable mod-wheel we just mentioned. The filter section is capable of high-pass, low-pass, notch and band-reject, which offers considerably more sophistication than your average low-pass-style filter. External sound sources can be routed through the filter section, which again provides another powerful string to the MS-20 bow.

The MS-20 is well known as a sonic all-rounder, equally at home generating fat or percussive bass sounds, leads, noise effects, along with crazy textures. And there are plenty of (traditional) MS-20 proponents including William Orbit, Aphex Twin, Air, Royksopp, and Jean-Michel Jarre.

PRICE $799 CONTACT CMI: (03) 9315 2244 or www.cmi.com.au

PROS It’s a real MS-20!

It’s analogue!

Cheap!

It’s a wee ripper!

Need I say more?

It’s difficult to decide where to begin with a review of Korg’s latest endowment to the synth-world: The MS-20 Mini. I don’t believe I’ve ever come across this situation. No synth manufacturer has gone to such extremes to recreate a 35-year old design, apart from perhaps, Moog, with its Minimoog Voyager Old School ‘Model D’ recreation.

That said, the Voyager Old School included functions not possible with the original Minimoog, so it wasn’t a strict rebuild.

And sure there’s been plenty of analogue modelling going on. Korg itself did a smashing job with the Korg Legacy Collection plug-ins and MS-20ic controller (more on this later), as it has also done with its analogue modelling synths such as the KingKorg I reviewed last issue. The

CONS No controller out via MIDI Some may be disappointed with tiny keys

other synthesis big-wig, Roland, has taken the path of paying homage to its vintage designs, with Juno, Jupiter and SH-101 style architecture and cosmetic regalia.

Then there are the plethora of boutique modular analogue synth designs from the likes of Doepfer, Vermona and ADDAC (to name but a few — there are dozens more).

These are hybrids in some respects, fusing 50-year old technology into modern variations. But as I mentioned, no manufacturer has gone to the extreme of rebuilding a synth to the original analogue specification — until now. Trust Korg to completely embrace its heritage. So I hope you understand my indecisiveness with this review, because in many ways this article could easily unfold as a review of the original Korg MS-20.

SUMMARY A real analogue MS-20 — just a little smaller and much cheaper. The perfect synth to whet your analogue appetite if you don’t already own one.

POOR MAN’S KORG

Most synthesis-heads would be aware of the genuine Korg MS-20 released back in 1978. In the day it was known as a poor man’s Moog, however, the MS-20 has since become part of analogue folklore, equally as collectable (albeit a third of the market value) as the Minimoog it was once seen as poor cousin to. So what made the MS-20 so popular? Well, to begin with it was semi-modular. In other words, alongside a set of onboard controls for the synthesis section, the MS-20 included a patchbay for shunting various modulation sources to various destinations.

This increased the versatility of the unit no end, allowing the synth to be controlled by analogue sequencers such as Korg’s SQ-10 or to control modular synths such as the Korg MS-50, a keyboard-less single-VCO design with a bucketload more patching options. (The MS-50 also purportedly ‘sounded better’ than the MS-20. I couldn’t tell you one way or the other as MS-50s are now rarer than the proverbial rocking-horse doo-doo. Expect to pay three to four grand if you do find one for sale.)

But I’m getting off track here. The MS-20 Mini (and the MS-20) is quite a synthesis workhorse.

It’s the type of instrument you can continually turn to, churn up a sound quickly, sample it and get back to composing. This is how many would use the original unit, otherwise you’d need a MIDI-to-control voltage converter to get the unit triggering from your DAW. Fortunately the new MS-20 Mini features a MIDI input and a USB port. The vanilla five-pin MIDI input will allow triggering notes from the MS-20 Mini, and hooking the unit directly to your DAW machine via USB will enable bidirectional MIDI control.

In other words you can use the keyboard of the MS-20 Mini to play notes into your DAW/ sequencer.

MINIATURE REPLICA

Exactly why Korg decided to re-release the MS20 at 86% of its original size is a bit of history in itself. In 2007, Korg released a software suite of instrument plug-ins — the Korg Legacy Collection. These plug-ins replicated classic Korg synths such as the MS-2000, Poly Six, M1, Wavestation, and of course, the MS-20. For a limited time you could buy the package with an MS-20 style controller — the MS-20ic. The controller was an 86% size replica of the MS-20 with mini-sized keys and 1/8-inch patchbay jacks (rather than the original MS-20’s 1/4-inch patchbay jacks). In order to function as a plug-in controller, the MS-20ic sent MIDI information from every knob on its front panel to the Korg Legacy MS-20 plug-in — even to the point whereby plugging physical patch leads into the controller’s patchbay replicated on-screen patch cables in the plug-in’s GUI. It’s quite a clever little unit, and also controls Korg’s iMS-20 application for iPad, which, I might add, includes a virtual SQ-10 sequencer.

As it transpires, the MS-20 Mini is similar to the controller — similar reduction in size, same 1 /8-inch patchbay jacks — but with real analogue

guts. Some might complain about the mini-sized keyboard, but I’m personally a fan of mini-size keys — it takes me back to the days of belting out bass-lines on a Yamaha DX100. The only other, very slight niggle, is the signal out jack is a small 1 /8-inch jack. But this is so you can easily route the output through the unit’s ‘external signal processor’ using the supplied 1/8-inch patch cables if needs dictate. The MS-20 Mini comes with 10 of these patch cables, and they’re yellow — just like the virtual patch cables in Korg’s software variations of the MS-20.

MANUAL PATCH

The unit also comes with a reprint of the original MS-20 manual which is kinda cute — this is, after all, an MS-20. The manual includes a few ‘patch sheets’ to get you started with ‘classic’ synthesiser sounds such as ‘Steam Locomotive’ and ‘UFO’ (‘Funny Cat’ didn’t make the cut). One of the interesting patches laid out in the manual makes use of patching a guitar into the unit. With four patch cables you can delve into the 1978 version of guitar synthesis. Prepare to alter your playing

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The MS-20 Mini shines like the classic synth it is

style somewhat as the results can be incredibly random, almost unusably random, just like the original, but it can be a lot of fun and a great source of sound bites and inspiration. Of course you aren’t restricted to inputting guitars through this patch — stick anything you like through it — bass, vocals, didgeridoo, wailing cats, it doesn’t matter. This is all voltage controlled stuff so anything you can get into a microphone or an amplifier (and out into the MS-20 Mini) will provide a suitable source. In fact, while scouring the interwebs for MS-20 information I came across an addendum to the MS-20 manual which explains some patch setups for external sources — we’ll upload this to the AT site for those interested.

Another awesome patching trick is the ability to send sounds straight into the MS-20 Mini’s filter section. Here you can use both high and low-pass filters — each with resonance — or ‘Peak’ as it’s labelled on the front panel. You could even use an expression pedal to alter the filters for ‘wah’-style effects. What you’ve got to remember here is the MS-20’s filter is renowned for its creamy sound, plus the fact that when the Peak (resonance) controls are right up the filter will oscillate wildly. You can achieve an incredible amount of variation from this filter set — and it’s all completely analogue. Hoorah!

When it comes to more traditional synthesis manoeuvres, the MS-20 Mini shines like the classic synth it is. Guttural bass sounds, screaming leads, and every quirk and warble in-between is possible with dual oscillators, dual filtering, and enough modulation options to keep you fumbling about with patch-leads for years to come. It’s a ‘go-to’ synth. There are, however, a few slight differences under the hood to the original MS-20. For starters there’s a newly designed VCA offering a better signal-to-noise ratio than the 1978 model, and the power supply is now an external adaptor. There’s also an auto power-off function which can be defeated by holding down some keys on the keyboard when powering up. Re-enabling auto power-off is a reversal of the same procedure. Other than that the design is apparently a ‘faithful reproduction’ of the MS-20 with MIDI I/O. HARD OR SOFT

I’m in a bit of a quandary as to whether the MS20 Mini is something I need to own. The price is phenomenal (an original will set you back $1500 plus) and yes, it’s a real MS-20 with real analogue circuitry. It sounds absolutely awesome as real analogue synths do. Putting the MS-20 Mini up alongside my Korg Legacy Collection MS-20 plug-in cast a lot of light upon my decisionmaking process as the sound from each is strikingly similar. I’ll admit the MS-20 Mini does have the edge but is it that much of an edge to warrant the expense? Do I want a real analogue MS-20 that is basically an old synth with MIDI under its belt, or do I stick with the plug-in with the option to save patches and record?

My humble question is, if Korg could make the MS-20ic controller and the MS-20 Mini, could it be possible to combine the two so the MS-20 Mini could send controller information from its knobs? Imagine the possibilities if the unit sent controller information. You could save patches to your sequencer, recall patches for each composition, and you could record some excellent realtime tweaking. If possible, it would make a killer resurrection even better than the original and still be far more affordable. The grapevine tells me Korg has further analogue surprises up its sleeve. I sincerely hope a hybrid MS-20 Mini, one offering both the analogue engine of this synth with the controllability of the MS-20ic is in the works.

There are undoubtedly many who’ll be all over the MS-20 Mini simply because it’s the real deal — a ridgy-didge, stand-alone analogue synth.

I wouldn’t blame them at all, and the lack of recall may be exactly what they’re after. Korg has been whetting the next generation’s appetite for analogue synthesis, with keen debuts like the miniature Monotron, Monotribe, and now, the Volca series. The price-point of the MS-20 Mini makes for a logical next step. It’s a solid purchase.

Go forth and squelch!

REVIEW

PROTOOLS 11

Avid has made its final migration to 64-bit, with faster everything, including faster-than-real-time bounce. And while there are huge upsides to upgrading, there are some tradeoffs to trading up.

NEED TO KNOW

Review: Brent Heber

PRICE ProTools 11 software (full version): $769 ProTools 10 to 11 upgrade: $329 ProTools 9 to 11 upgrade : $439 ProTools Express to Pro Tools 11 cross grade: $549 ProTools HD 10 to 11 upgrade: $659 ProTools HD 9 to 11 upgrade: $1099

CONTACT Avid: 1300 734 454 or www.avid.com

PROS Faster-than-real-time Offline Bounce Improved visual feedback in Mix window More powerful and efficient Native engine Headroom inside plug-in processing Avid MXF video can be played back without video hardware

CONS No longer supports HD TDM, Control 24, Pro Control, blue interfaces Dropping Complete Production Toolkit Some plug-ins won’t be available immediately, if ever No loudness or True Peak metering built in (yet?)

SUMMARY There are genuine advances across the board in ProTools version 11 as Avid completes its move to 64-bit. Faster-thanreal-time offline bounce, internal plug-in headroom, and a new engine that has ’Tools motoring faster than ever. ProTools 11 is an easy decision for adopters of the new Avid hardware — but owners of older Digidesign gear have a harder decision to make.

Late 2011, Avid released ProTools version 10 and with it a statement that it would be the last version to support the ageing TDM architecture, some control surfaces and older ‘blue’ generation interfaces. Many studios around the world have quite a bit of that gear and raised their fists in ire at the pro audio giant for ‘rendering their hardware obsolete overnight’. But more reasonable folk could see the writing had been on the wall for some time. Operating systems were moving to 64-bit, and host-based audio systems and plug-ins were transitioning to 64-bit float audio paths, neither of which would co-habit easily with the older DSP used on TDM rigs. Still, the overriding sentiment from ’Tools incumbents losing an old system was that the new one better be worth it.

So, why is 64-bit a big deal? 64-bit can refer to two different things — the operating system and the applications it hosts, and/or an audio path’s summing mechanism. ProTools has had a 64-bit floating point audio mixer since version 9, but TDM rigs couldn’t take advantage of it, only hostbased systems. The 64-bit headline for ProTools 11 is the migration to a 64-bit application, and with that transition we should expect large gains in processing speed and power.

In order to take advantage of this new architecture, lots of other preliminary work had to be done. New DSP cards needed to be developed that were 64-bit audio and processing compliant (the new HDX cards) and plug-ins ported from existing 24-bit versions to 32-bit float for ease of processing (the new AAX native and DSP plug-ins).

Consequently, version 10 was a transition period.

It gave developers a chance to program for AAX before D-day, and allowed studios to stage their upgrades if they felt the need. All this was setting the stage for ProTools 11 and its new Avid Audio Engine — AAE, as opposed to the now legacy ‘Digidesign’ DAE. STEAMING ON

The rub? The new Avid Audio Engine brings with it some major advances over TDM. For one, plug-ins have headroom, as they are now 32-bit float. Say you wanted to push TDM faders up a skerrick and then pull the signal back into line at your sub master. But just as you achieved the balance you’re after, your EQ plug-in clips.

You may have had 12dB of headroom over 0 on those faders, but the problem is, you can’t push over 0 inside 24-bit plug-ins. With the increased headroom of floating point, this ceiling goes away and with it a new way of reading clipping inside plug-ins — they now show orange, not red, which is reserved for real clipping.

AAE also brings all your RAM into the mix.

Version 10 introduced a new RAM cache — a 64-bit chunk of hidden code running alongside ProTools — to house audio for playback.

It dramatically decreased seek time while increasing system responsiveness — similar to playing data off an SSD. Under the new AAE regime, those internal limitations have been completely removed, giving plug-ins access to

plenty more RAM than the 4GB limit of 32-bit applications. This means virtual instruments hosted inside ProTools can access a much larger chunk of memory for sample storage and/or calculations, which no doubt makes composers much happier. ProTools 11 also increased the number of instrument tracks available to composers from 64 to 128 on the non-HD version, and 256 on HD. AAE introduces one other very smart piece of system design: If there’s no audio on the timeline feeding a native plug-in, then that plug-in is deemed inactive and removed from the hostbased processing pool. Often on a large mix, you have chunks of major activity interspersed with less busy sections. In earlier versions you would have to lock in processing power for the busiest times throughout your mix, as the inactive state of a plug-in couldn’t be automated. Watching the system usage window in version 11, you can clearly see the flux as tracks come online for processing and then drop away again as they fall silent, leaving you with extra processing headroom across the board. System usage also now shows all the host-based processors (real and virtual/hyper-threaded) on your machine for improved monitoring of the system’s available native power.

Another change is the automatic delay compensation (ADC) engine in AAE is fixed at the 16,000 sample setting, the ‘Maximum’ on previous versions — there isn’t a short or long choice in your playback engine panel anymore.

There is a new option for ADC on side chains as well. Also, Playback Buffer settings no longer need to be juggled as often: all record armed tracks or MIDI tracks now operate permanently in the ‘ultra low latency’ domain designated, as before, by the playback buffer. The key difference here is that all other tracks will run at a 1024 sample buffer all the time, even when tracking. It will provide more grunt, with ProTools 11 doing all the CPU juggling within its ADC engine for you (this is not to be confused with the ‘Low Latency Monitoring’ mode, where plug-ins are automatically bypassed during tracking).

“

All this work; new DSP cards, new audio engine and new plug-in format, leads to one of the biggest feature requests Avid ever heard — faster-thanreal-time bounce… a reality in version 11

”

LET’S GET REAL TIME

With ProTools audio now being processed at 32- and 64-bit float, the application being 64-bit float and the OS being 64-bit as well, can you see the ducks lining up? All this work, new DSP cards, new audio engine and new plug-in format leads to one of the biggest feature requests Avid ever heard — faster than real-time bounce. A reality in version 11, you can now bounce your mixes in less time than it takes to play them out — significantly so — and more importantly they will nearly phase cancel with a real-time play out. Why nearly? Many reverbs and effects rely on a certain element of randomisation to generate their sound, so depending on processing choices, even two real-time bounces rarely cancel out completely.

In real world terms, a ProTools 11 ‘faster-thanreal-time bounce’ is identical to a legacy real-time play out, with no quality loss. Not only that,

Faster-than-real-time bounce is now a reality in ProTools 11, and you can even bounce straight from any bus.

but the HD version will allow multiple bounces simultaneously, great for exporting stems. Both HD and non-HD versions have a new tick box in the bounce window to generate an MP3 at the same time as your mix (which is a thoughtful addition for client approvals) and the bounce menu can now be accessed by right clicking on any bus you want to bounce out. The speed of the ‘offline’ bounce is dependent on CPU speed and the amount of the session loaded into RAM also has a distinct effect. From what I’ve seen, a feature film mix and its adjoining stems took about five minutes on a Nehalem Mac Pro with plenty of RAM, as opposed to 90 minutes real-time — a massive time saver!

COMPOSED PLAYBACK

Aside from these dramatic changes in the audio engine, Avid has also integrated its Media Composer-developed video engine directly into ProTools 11. This means that in addition to opening all the Quicktime file formats you’re used to, you can open a video editor’s MXF files directly, speeding up handovers. Overall, sync with the new video engine is much tighter than previous versions that relied on Apple Quicktime for playback and MXF files look great both on the desktop and using video hardware from Avid, Black Magic Designs and the newly supported AJA. If you’re running low on host power, you can even switch your picture quality down to ‘best performance’ to get that little bit more from your system.

Avid is also doing its level best to mix-nmatch Media Composer and ProTools in other ways, with a consistency in jargon and visual appearance that renders version 11 a bit darker than previous versions. I’m not sure if I’m a fan of the new look, it could prove a bit harder on the eyes long term, but in a dark environment it may be gentler — hard to know without solid use.

ProTools 11’s new look metering includes VU, K-system, and a variety of PPM standards, and the new gain reduction meter can show the total gain reduction of each channel.

METERS JUST SHY OF TRUE PEAK

The biggest visual differences are in the Mix window. The faders now throw 30% longer and the meters are strikingly updated with a bunch of different metering standards implemented to suit the various geographies and industries that use the DAW. PPM meters of all flavours are there and VUs of various iterations also. I quite like the feel of the Bob Katz K-Metering formats, showing both peaks and RMS values simultaneously, and comprehensive preferences allow you to customise settings like where the colour breaks reside. You can also get to your metering preferences simply by right-clicking on them in the mix window. Options noticeably absent, however, are the ‘new’ True Peak and loudness standards. Given Avid’s reliance on the broadcast sector in Europe and the US (both requiring stringent loudness compliance) I’m scratching my head as to why it would overlook such a fundamental requirement, especially since Nuendo has had these features for some time and Avid’s own Pro Limiter has comprehensive loudness metering built in. While this seems like an oversight, the overall meter upgrades are very welcome, and the specific varieties seem to have been chosen based on what might be required per channel, not bus.

Keeping with the metering trend, sends now have tiny meters built in, too small for any real use, but useful as a ‘signal present’ indicator. They also more easily expand to show faders, pre/post state, slightly larger metering and a bypass button without stealing access to other sends.

ProTools 11 HD has the biggest metering change however, in that any compressors or expanders on a channel can now show their gain reduction on a full size permanent meter alongside the main channel meter in the mix window. This is a fantastic change for the better and I’m sure many engineers will find it hard to move back to earlier versions after working with this visual feedback for a few hours. Of course, this is dependent on the plug-in manufacturer implementing support for it, so if you have a favourite third party dynamics plug-in be sure to chase the manufacturer about it before it releases its AAX 64-bit version (McDSP and Plugin Alliance have it working nicely).

ProTools 11 allows automation to be recorded while recording audio.

WORKMANLIKE FEATURES

Although the flagship features of version 11 are the offline bounce, 64bit power and metering overhaul there are a bunch of other workflow enhancements. One of the long-standing limitations of ProTools when recording was disabled automation. It’s now removed, which is particularly useful when recording concerts or for new Dolby Atmos workflows.

Automation is also now sample accurate, whereas in the past it may have slipped around a little as your CPU was pushed. It’s also one of the reasons Avid can achieve true phase-cancelling faster-than-real-time bounce, and consequently, why ProTools doesn’t cater for VST or AudioUnits. The AAX code was written to include sample accurate automation tracking, something VST and AU hosts are still grappling with.

If you monitor your mixes quite loud, there’s also a new Fade In pref to set an automatic fade up duration on starting playback — avoiding full scale instant woofer excursion.

The Workspace browser has also had a major upgrade. The multiplicity of similar-looking browsers has been merged into a single browser window, unifying all the previously separated functions into an overhauled database engine that runs quicker and with more stability.

Importantly, with the end of RTAS and TDM plug-ins and Waves choosing not to develop AAX DSP, there are many of us who may want to delay upgrading our hardware. To make this dilemma less of an issue, ProTools 11 is the first version that can be co-installed alongside a working version 10 build — that’s right, you can run 10 and 11 on the same operating system (because they are such different applications). The demo HDX rig I was working on happily shut down 10, opened 11, and then reopened 10 without a stumble, making adoption much easier as you don’t lose easy access to any RTAS plug-ins that have yet to migrate over to AAX.

MAKING THE AAX CUT Considering making the jump but unsure of which plug-ins have made the AAX leap? Avid has compiled a list of those already migrated to the new format at www.avid.com/plugins. Version 10 was supposed to be a transition period for plug-in developers, but there are still many playing catch up. For instance, Steven Massey of Massey Plugins has been working on AAX versions, but no release date has been set.

The native system usage in ProTools 11 is dramatically lower on the same computer compared to version 10.

ARE WE COMPATIBLE?

ProTools 11 works in the .ptx document format, same as version 10, and projects can be moved between the two without loss of data for the most part: plug-in settings, however, may not always carry between the two versions so be very careful! Some developers have to re-code from the ground up for AAX making totally new plug-ins, so their TDM/RTAS settings files cannot be opened in the newer versions. Double check this workflow before relying on it. Avid is a culprit of this as well, with its Mod Delay III AAX not opening presets from the dropped Digirack delay plugins, breaking backwards compatibility.

AAX plug-ins are also confusingly different in version 10 and 11, due to the 32- vs 64-bit disparity. So even though you may have plug-ins ported to AAX for version 10, they won’t work in 11 unless they are 64-bit capable, something many developers are furiously working on right now. Some of Avid’s own plug-ins haven’t made it over to 11 either, notably its TCE and Pitch plug-ins, which are hinted still to come. Avid’s TL Space and Revibe reverbs, and Phasescope/Surroundscope are also yet to port to 11.

THE LEAP OF FAITH

ProTools 11 is certainly a big step forward for Avid, as far as system architecture goes. How optimised this new engine is and how well it performs is difficult to say without more third party plug-ins in the 64-bit AAX format. Certainly, comparing strictly Avid plug-in sessions between versions 10 and 11 you see a dramatic drop in host-based system usage in the newer version. A film premix on my old native rig opened at 24% on version 10, yet shows around 8% CPU usage on 11. Given Apple’s decision to drop PCIe cards from its new Mac Pro design, these sorts of numbers tell me that by upgrading to version 11 I can get a few more years from my current Mac Pro, an important factor in the current economy especially given how the

media production landscape is changing and budgets keep shrinking. However, that only makes sense if your plug-ins of choice make the transition to the new 64-bit AAX format (see AAX breakout boxes for more info).

Looking at performance across the application, it seems strange that some aspects of processing haven’t been sped up by the transition to 64-bit. Rendering Audiosuite plug-ins seems just as lethargic on version 11 as it does on 10, and sample rate conversion of files also didn’t seem to benefit from the change either — both processes steal significant chunks of my studio time and I was hoping to see some improvements. Hopefully it’s just a sign of further 64-bit optimisation to come.

Whilst some parts of ProTools seem untouched, the vast majority of the system feels significantly fresher on 11; the new darker GUI, the metering enhancements and the snappy performance of the new engine make for an impressive update that is much more than simply offline bounce capability.

REVIEW

ARTURIA SPARKLE

RHYTHM PROGRAMMER & CONTROLLER

Arturia trims the excess off Spark to let the good stuff shine through in SparkLE.

NEED TO KNOW

Review: Brad Watts

PRICE $329

PROS Great size

CONTACT CMI: (03) 9315 2244 or sales@cmi.com.au

Easy to use and play Great and diverse library of sounds Well made

CONS Simply a controller

SUMMARY While products badged ‘LE’ typically are some bastardised sawn-off version of an original, SparkLE is everything good about Arturia’s Spark rhythm controller made more convenient.

The drum machine. The format has fallen in and out of mainstream favour over the last couple of decades. During the 1980s and ’90s, these devices were a mainstay of all electronic musicians’ equipment lists. Yet come the 21st century, the drum machine faded into obscurity as computer-based DAWs and plug-ins took over.

In some ways this was a shame, as hands-on rhythm programming can be an incredibly immediate tool in its own right. Arturia is one manufacturer that’s acutely aware of this avenue of beat production, and served the electronica community handsomely with the Spark ‘beat production centre’ in 2011. Spark is a hybrid design consisting of a tactile control surface tethered to computer-based sound generation and programming software. The best of both worlds in many ways: the breadth of software synthesis and recall combined with a drum machine-style controller.

However, the original Spark controller was quite a large beast. At 36 by 27cm, it took up a huge chunk of desktop studio space. And in a world where bedroom producers are squeezing themselves into ever smaller creative caves, I’d imagine many asked for a compact version of the Spark control surface — enter the endearingly monikered, SparkLE. THE NOT SO LIGHT

Traditionally, the ‘LE’ suffix usually implies a sawn-off version of software or computer-related hardware. ‘Light Edition’ is the typical expansion of the acronym. While the SparkLE controller may indeed be a sawn-off version of the first Spark controller, I don’t believe it’s missing anything a percussion controller surface requires.

As mentioned, the SparkLE controller is petite in comparison to the Spark controller. It sits flat on your desktop or lap and measures a mere 28 by 17cm, and sits a meagre 17mm high (plus knob height). It’s a nice size to have alongside your laptop or computer keyboard, and quite suitable for taking with you in a backpack. It even comes with a neoprene, flock-lined, carry bag for onroad protection. Very tidy.

Across the lower section of the SparkLE top panel are eight velocity sensitive pads accessing two banks of sounds. They do the trick, and light up blue when you hit them. Sweet! Up from here are the typical 16-step buttons for step programming drum patterns, just as you’d find on any oldschool rhythm machine. This doesn’t restrict the unit to 16 steps per bar, as pattern length can be extended to 64 steps when required.

Over to the right of the unit are three assignable instrument parameter editing pots, then to the left is a large knob for choosing Project, Kit, and Instrument presets. To the top are dedicated volume and tempo knobs, and below these is an XY touchpad. This controller device is gold, and with it you can affect filters over either individual instruments or an entire kit. It can

also be used to trigger a ‘Roller’ effect whereby the pressed instrument pad can be forced to repeat at particular bar divisions, and a ‘Slicer’ function slices a pattern and forces the pattern to repeat from 1/64th through to ¼ sections of a pattern — kinda neat for live performance or those with an Attention Deficit Disorder. Again in the performance department are pad, mute and solo buttons and record, stop, and play/ pause buttons. The controller can also be bumped into MIDI mode where each button and knob can be reassigned MIDI output duties so you may control other MIDI devices or software. All this functionality (and more) is embodied in a controller that feels as though it will take a right thrashing for years to come.

THE HEAVY

On the surface, as it were, SparkLE seems pretty straightforward — it’s a rhythm programmer.

How complicated can it be right? Well, let me assure you there’s a whole lot more going on under the bonnet in the Spark software.

This is the same software you’d acquire if you bought the bigger Spark controller and it’s quite powerful. Firstly, the sound library is immense, with sample and audio concoctions covering just about any drum and percussion sound you can imagine or recall. The software includes sample playback and layering, along with Arturia’s virtual analogue engine, and a physical modelling engine. That’s a ship-load of sound generation options completely at your disposal.

Samples include a stack of vintage recreations such as the Roland TR lineage, and Simmons kits, along with LinnDrum, Drumtraks and Oberheim DMX-style sounds. Then there’s

house, dubstep, techno, hip-hop, and r ‘n’ b-style kits (if pigeon-holing sounds is your bag) and a bucketload of acoustic style kits.

The editing possibilities are impressive, with pitch, panning, decay, attack, and filter control (with resonance) over each sound. Equally impressive is compliance with other systems. You can export patterns as MIDI files or audio files.

Plus, 16 separate outputs are supported should you have the I/O. When you come to combine all this power, there’s intrinsic control over pattern and song compilation, and a rudimentary mixer with access to a pile of insert effects and two sends from each channel through to return effects. The software is supplied as a stand-alone program, or as plug-ins for Audio Unit, VST, RTAS-based systems, with Avid’s AAX protocol in the pipeline.

THE SPARK

I think the deciding factor as to whether SparkLE is for you, is if you’re happy having your rhythm programmer tethered to a computer.

Some may prefer their rhythm programmer to be self-contained. But that said, the Spark software itself is actually worth owning simply for the sound creation possibilities and the quality of the sounds — the controller is really the tasty icing on an excellently prepared cake.

As a performance device I believe Arturia has sacrificed just the right amount of accessibility (in terms of tactile buttons and knobs) while providing a friendly rhythm programming interface. I’m sure the format will inspire beats producers in exactly the way ‘drum machines’ always have.

REVIEW

SCL-1 ERGONOMICS

The SCL-1 has toggle switches for hard-wire bypass, stereo and dual mono operation modes and another switch to select which channel feeds the large backlit gain reduction VU meter. All pots are continuously variable. Separate input and output level controls remain autonomous even when in linked stereo mode, allowing for subtle L/R level matching. In stereo mode all other stereo parameters are controlled by the Channel 1 controls. The SCL-1 attack time can be set to a lightning quick 100us when on ‘10’. Compression ‘Slope’ ranges from 1:1 to 20:1.

CHARTER OAK

SCL-1 COMPRESSOR

A compressor that crosses the threshold twice without leaving a trace, just don’t ask it to pump.

NEED TO KNOW

Story: Greg Walker

PRICE $3599 CONTACT Galactic Music: (08) 9204 7555 or info@galacticmusic.com.au

PROS Great sonic flexibility and transparency Unique and very effective design topology High headroom and build quality

CONS Expensive Doesn’t do compression effects

SUMMARY A quality compressor that excels at program compression with minimal artefacts.

The SCL-1’s Static and Dynamic Threshold controls allow for subtle tweaking of the compression behaviour while the unit’s high headroom ensures transparent results.

If there was ever to be a Jedi-style school for audio ninjas this would surely be one of their mantras — ‘With great dynamics control comes great responsibility.’

I can see a robed padawan quietly meditating before a pair of giant futuristic VU meters and then weighing his choice, hand hovering over a bank of outboard compressors… What shall it be this time? The API? The Al Smart? Perhaps the Amek? What of the ELI Fatso, the JML MAC or the Tube-Tech? A dozen sonic blueprints present themselves, with both desirable and undesirable compression artefacts, transient responses and tonal characteristics. In a bold move, the apprentice’s hand rests upon the controls of an unheralded contender, one that doesn’t claim to colour or drive the sound in any particular direction but offers an astonishingly transparent path to dynamics control with no unpleasant side-effects or tonal compromise — the Charter Oak SCL-1.

DEMING ON REAL DYNAMICS

SOFT LANDING

With the SCL-1, Charter Oak’s head honcho Mike Deming clearly set out to honour the audio ninja’s compression mantra. His self-imposed task was to build a stereo compressor capable of handling mix bus duties without introducing all the artefacts we often associate with generously applied program compression — pumping, rush-up after rests, transient smearing and distortion et al. In order to succeed at such a perilous mission the SCL-1 utilises an unusual topology built around a parabolic average charge curve in the rectifier circuit. Using the Dynamic Threshold control to set the circuit’s sensitivity, the threshold varies according to not only the incoming signal but also the capacitor charge held over from the previous moments of musical input — effectively creating a continuously variable ‘soft landing’ back to average program level.

In addition, the SCL-1 provides a separate Static Threshold control that is used to raise or lower the control voltage and thus control the amount of swing between peak and average levels — acting as a hard/soft knee control.

Using these two controls and tweaking the attack and release settings it is possible to grab and control big transients in an effective way while the body of the mix sails through on its own more gentle dynamic trajectory. While I can hear the dance fraternity groan as whole mixes fail to duck momentarily after each kick drum wallop, there are many styles of music where this kind of transparent dynamic control is an absolute godsend. For acoustic and classical mixes, indeed any mix situation where no extra colouration or ‘compression effect’ is required, the Charter Oak brand of dynamic control has undeniable benefits.

ALL CLEAR

My first experiences with the SCL-1 were on some fairly complex and dynamic folk-rock material. I followed the manual’s instructions and first calibrated the outputs of my converters to the unit by running reasonably loud program material at +4dB output levels. Charter Oak

Mike Deming: “In my 22 years of producing records one of my biggest problems, especially with acoustic and quieter musical styles, was trying to find a compressor we could put across the mix bus that was free of artefacts. Rush-up was my worst problem, I would have to cut my mixes into pieces to readjust my release curves and then stick them back together afterwards. You set your release time and it’s a time constant in every compressor design in the world except the SCL-1. Unless the band is playing in exactly the same tempo from start to finish the result is not going to be good, especially when there’s a rest or gap in the music.

“The way we were able to get the rectifier circuit to work in the SCL-1, it’s like it has a mind of its own. You use the settings to select a range of dynamic compression. The bottom is going to be the constantly changing average RMS, which is where the unit always returns to, and the top is the amount of dBs of compression you’re applying on peaks.

“

I found it hard to make this compressor behave in anything but a very musical way

”

recommends that the Static Threshold control is set at seven and left until the user is happy with all other aspects of the compression chain.

Once this happens the control can be tweaked in the manner of a hard/soft knee control to fine tune things at the end of the process. By its very nature, a transparent dynamics controller makes the adjustment and tuning of attack, release and sensitivity thresholds a more subtle task and I found I needed to concentrate and really use my ears to hear all the nuances. After a bit of time and experimentation with the device I was achieving extremely pleasing results with expanded detail and RMS power in the body of the mix and effortless control of more unruly transients. There was also a definite smoothing of the whole soundstage and a sense of solidity about the mixes that to my mind showed the benefits of the unit’s all-analogue design. Using the attack and release controls allowed for some very nuanced shaping of the rhythmic content and, like a lot of truly high-end gear, I found it

“With all the other compressors we’ve had here in the studio over the years we were only ever able to get a dB or two of compression on the meter before it started to collapse a mix, but this thing is quite different. We even tell you in the manual to apply between 3 and 7dB of compression because that’s where you get maximum modulation of the signal and a 4dB slope shift. Right at that point the mix starts to sound glued together and really nice.

With this box, the more you make the meter bounce, the better it sounds. It’s kind of the opposite to a lot of other compressors.

“The key to how the SCL-1 behaves is all to do with the way the capacitors charge and discharge in the control circuit.

It’s actually dependent on two things; what just happened and what’s happening at the moment. It will automatically give you a quick release on staccato notes and a nice long release on legato notes within the range that you’ve selected.”

hard to make this compressor behave in anything but a very musical way.

SOME LIKE IT HOT

It is worth noting that the SCL-1 loves a hot signal and can effortlessly cope with 10dB or more of gain reduction. The sweet spot seems to be at around 5dB of gain reduction but there’s no need to stop there, the over-designed component tolerances, sturdy PSU and transformer outputs are more than equal to the task of staying on target regardless of how much juice is applied.

Moving to more of a mastering-style chain, I hooked up some quality EQs pre and post compression and did some tweaking on some heavier musical styles as well as some gentler ambient tracks. Again the SCL-1 came up trumps with its ability to fine-tune the transient response and the relationship between peak and average dynamic levels. Again I was impressed by the tonal transparency of the mixes even at heavy compression settings. There really was no noticeable change in tops, mids or bottom end and this of course explains the lack of the increasingly ubiquitous high-pass filter circuit — it simply isn’t required. Finally I switched from stereo to dual mono mode and ran vocals and electric bass through the unit, once again coming away very pleased with the results. No matter what I threw at it or how hard I drove the unit, the result were never less than sonically pleasing and I’ve got to admit to being very impressed by Charter Oak’s offering. The SCL-1 is undeniably a transparent high-end device that delivers on its promise of minimal artefact compression.

REGULARS

LAST WORD with

ROB PAPEN

At age 15 I saved up for a Korg MS-20 and an SQ10 sequencer. The year was 1979. The MS-20 was one of the first affordable synths. The Arp 2600 or Minimoog were beyond my reach.

level of expression. Everyone dropped their analogue synths at that point. If you played in a Top 40-style band, everyone wanted a DX7, so you could pick up a secondhand Jupiter 8 or Oberheim for peanuts.

Growing up, I took home organ lessons. I hated the organ with a passion, but it was a good grounding.

The best part of my job is knowing when a sound is ready. I hear it and that’s the rush I get. If I hear it again on the radio in someone else’s mix, that’s cool as well, but not as much of a buzz as when I know I’ve got a sound ready.

Listening to Tangerine Dream, Jean Michel Jarre and Giorgio Moroder, I was totally intrigued by electronic music.

Rob Papen has a large and loyal following with his eponymous virtual instrument company (www.robpapen. com) but initially came to prominence as probably the first celebrity synth soundset programmer.

A Rob Papen preset is a special thing.

My first appearance as an artist was at a Klaus Schulze fanclub day. I met two other guys who had a studio and were doing electronic music. We started a band in the early ’80s. We made a couple of very successful underground electronic albums. We even had a No. 1 hit in the Benelux countries with a spinoff of the band, called Nova. I was 18.

While I was in the band the sounds I designed were for us only — I didn’t share. Until I heard the Waldorf Microwave in 1989.

I was really excited about that synth, as it was a rackmount version of The Wave from PPG. But when I heard the Microwave I was disappointed with the presets. I dropped them a line: ‘How about I make some sounds for you?’, which became the Signature Sound Card. The Techno Card followed and was a huge success for them.

Alesis Andromeda: That’s a pain to program. It’s so complex.

Each stage of the envelope can have its own curve. You almost have to have a default preset to make a preset. The Andromeda looks fantastic, sounds good, but it’s not a synth that matches up to the Jupiter 8. The Jupiter 8 is far simpler, but ‘far simpler’ is often more rock ’n’ roll and friendlier.

People think I’m a maths guy, but I’m more instinctive. I know how things work but I’m more intuitive.

E-mu Samplers: I purchased an Emulator in the ’80s and did sounds off my own bat. I sent them to E-mu to see if they were interested in reselling them. They loved them. The E-mu samplers had a great filter and a very nice algorithm if you detuned or transposed a sample. I had a lot of fun doing sounds for the E4 and E6400.

After that I designed sounds for a whole host of different synths, for Waldorf, E-mu, Ensoniq and others. My work on the original Access Virus really elevated my name [all the sounds initialed ‘RP’ are Rob’s], and my final third-party sound set was for the Alesis Andromeda.

I started out with a with a one-voice synthesiser and at that time it was a curse — it was a race to get more polyphony and to get multitimbrality. But now the curse is being totally overwhelmed by the range of sounds — you can easily drown. It’s not bad to limit yourself. Limits help you be creative.

In the late ’90s I teamed up with John Ayres — the programming brains behind our company — and formed Rob Papen Instruments. We have nine products in all, mostly soft synths — including Predator and Blade — along with three FX plug-ins. I’m the artist and synth geek, that’s what I bring to Papen instruments.

E-mu or Ensoniq? That’s a hard one. I did two sound sets for the Ensoniq ASR10, which had a very distinctive sound. It’s a very powerful sampler, even now. Ensoniq and E-mu are the top sampler brands. They set the benchmark for me.

Waldorf Microwave: It had a wavetable and an analogue filter and analogue VCA, so it was a very distinctive sound. It could sound awfully harsh but you could produce sounds totally different to a regular subtractive synth.

I think the biggest asset of my sounds is how musical they are.

You can have a sound that is nice to listen to in isolation but you can’t make music with it because it doesn’t fit with anything. My sounds, although not always spectacular, are very usable and solid as a rock. And as a remixer or producer, when you don’t have time to build everything from scratch, that’s what you need.

Yamaha DX7: When the DX7 came a long, I thought, ‘what is this crap?!’ I had a Jupiter 8 and the DX7 was a real shock for those accustomed to dials and sliders. But the sounds were very different, and special. I have a huge respect for those who developed the original soundset, because it was painfully hard to program. It was velocity sensitive, so the DX7 brought a new

Software piracy? We now rely on people registering their purchase with us, and that guarantees support, updates and new sounds as they’re released. It’s amazing how shameless people can be. They’ll log-in their virus code and expect support! It’s frustrating. When I was a teenager I had to do a paper round for years to save up for a Korg MS-20. I didn’t go to the shop and steal it. Piracy is an attitude.

The most famous synth preset? Perhaps the Shakahashi sound — the flute from Peter Gabriel. The Yamaha TX81 bass sound was also used on countless tracks.

My synth Top 10? Well, you’d have the Roland Jupiter 8 in there, the Waldorf Microwave, the Sequential Circuits Prophet 600 for its brutal sound, the Oberheim OB8, Access Virus, the MS-20 for sentimental reasons, the E-mu E4 and Ensoniq ASR10, and the Yamaha CS-80 (although the closest I’ve come to playing it is the CS-60)…

But only one synth can sit on the throne and that’s the Minimoog. It has such a distinguished sound.

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