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Editor Mark Davie Publisher Philip Spencer Editorial Director Christopher Holder Editorial Assistant Preshan John Art Director Dominic Carey

Regular Contributors Martin Walker Paul Tingen Guy Harrison Greg Walker Greg Simmons Blair Joscelyne Mark Woods Chris Braun Robert Clark Andrew Bencina Brent Heber Anthony Garvin Ewan McDonald

Graphic Designer Daniel Howard Advertising Philip Spencer Accounts Jaedd Asthana Subscriptions Miriam Mulcahy

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AudioTechnology magazine (ISSN 1440-2432) is published by Alchemedia Publishing Pty Ltd (ABN 34 074 431 628). Contact (Advertising, Subscriptions) T: +61 2 9986 1188 PO Box 6216, Frenchs Forest NSW 2086, Australia. Contact (Editorial) T: +61 3 5331 4949 PO Box 295, Ballarat VIC 3353, Australia.

All material in this magazine is copyright Š 2016 Alchemedia Publishing Pty Ltd. Apart from any fair dealing permitted under the Copyright Act, no part may be reproduced by any process with out written permission. The publishers believe all information supplied in this magazine to be correct at the time of publication. They are not in a position to make a guarantee to this effect and accept no liability in the event of any information proving inaccurate. After investigation and to the best of our knowledge and belief, prices, addresses and phone numbers were up to date at the time of publication. It is not possible for the publishers to ensure that advertisements appearing in this publication comply with the Trade Practices Act, 1974. The responsibility is on the person, company or advertising agency submitting or directing the advertisement for publication. The publishers cannot be held responsible for any errors or omissions, although every endeavour has been made to ensure complete accuracy. 01/03/2016.

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Interface Roundup




Richards Rolls On: Making Crosseyed Heart

Cooked Raw: Recording at Welcome To 1979


Kurt Vile Hits Sonic Byways to Avoid the Ticking Clock

How To: 5-Hour Progressive House with 44 Project 46 and FL Studio 12

Soyuz SU 011 & SU 017 Tube Microphones AT 6



From Modding to Modelling: 40 UBK’s Gregory Scott

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Forward-thinking line-array design starts here. The result of rigorous R&D, the introduction of the new X-Line Advance family sees Electro-Voice push the parameters of line-array performance to the next level. X-Line Advance utilizes state-of-the-art EVengineered components and incorporates a range of innovative new features, all of which work together to surpass the capabilities of other line arrays, and all in a significantly more compact, flexible, and quickerto-set-up package.

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Designed, engineered, and tested for ultimate reliability by Electro-Voice in the USA. Learn more at:

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1300 026 724 AT 7








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RACK-READY REDDI REDDI has made a name for itself as a DI that sounds as solid as its heavy red chassis. Now A-Designs has launched the rack-mountable REDDI V2 — a dual-mono-channel, 19-inch, 2U model. The REDDI V2 is not a high-gain device. Its gain structure and design is all about avoiding the slightest compromise in sound quality. Inspired by the sound of the Ampeg B-15 tube bass amp, the REDDI V2’s 6N1P-driven amplifier feeds signal directly into a custom Cinemag output transformer to provide a harmonically rich tone. The REDDI V2’s wide frequency range (20Hz - 60kHz) helps prevent

in-band phase shift, resulting in outstanding detail and realism. The distinctive bright red steel chassis sports two front-panel Neutrik XLR/¼-inch combo input jacks, two balanced XLR outputs and two ¼-inch Thru-puts, for sending signal to a bass amp. A 0-16dB Level control and red LED power indicator complete the front panel controls, while the back panel contains a heavy-duty toggle switch for ground lift, along with the IEC power outlet. Audio Chocolate: (03) 9813 5877 or

DIGIGRID DESKTOP SERIES Digigrid’s cute new Desktop series consists of four units: the [Q] Headphone amplifier, [D] Desktop interface, [M] Musician recording interface, and [S] PoE Switch. The [Q] Headphone Amplifier is an audio interface with 1/4-inch and 3.5mm outputs and four input options (XLR, CAT5, analogue and Bluetooth). The smart little 90mm-cubed box is designed to drive headphones at serious volume, and is fully controllable to suit any audio application where high-end audio is required. Coupled with an AES/ EBU interface, it allows a direct digital interface loud enough for any drummer. The [M] Musician recording interface has two inputs and two outputs, again, at only 90mm cubed.

It’s designed for plug and play simplicity, with simultaneous monitoring and recording capabilities through two dedicated I/O. There’s also an optional mic-stand adapter plate, which works with both [M] and [Q] Digigrid [D] is the more expansive option, with four inputs (two mic/line, two line/instrument) and six outputs. There are two outputs with level control and two further fixed level line outputs. Last is the sleek and compact Digigrid [S], also Dante compatible, featuring one upstream (nonPoE) port, and four downstream PoE ports, which allows connection to four further Digigrid devices. Group Technologies: (03) 9354 9133 or

ARTURIA MATRIXBRUTE Arturia has extended its gnarly range of synths with the Matrix Brute. Dubbed an ‘avant-garde synth’, it’s a completely analogue synthesiser capable of fully programmable monophonic/duophonic tones. It has three oscillators which share the trademark of its other ‘Brute’ siblings, Steiner-Parker and ladder filters, three ultra-fast envelope generators and quality analogue effects. The ‘Matrix’ part of its name comes from the modulation matrix at the heart of MatrixBrute, making the most of each

‘module’ through a matrix that offers thousands of potential modulation routings. But unlike an old-school modular system, the MatrixBrute lets you save each patch you make. The matrix can also function as a 64-step sequencer with separate Step, Accent, Slide and Modulation options for the sound to evolve as you want it to. Take one look at it, and you’ll know the MatrixBrute means business. CMI Music & Audio: (03) 9315 2244 or

SHURE’S DUAL DIAPHRAGM KSM8 Shure has released the KSM8 Dualdyne cardioid dynamic mic — the world’s first dual-diaphragm dynamic handheld microphone for vocals. The KSM8’s retro appearance is a nod to Shure’s rich history of microphone production and expertise. But what’s impressive is its ability to provide virtually no proximity effect and outstanding off-axis rejection, thanks to the patented Dualdyne cartridge that features two ultra-thin diaphragms — one active and one passive — and an inverted airflow system at the heart of the KSM8’s design. Shure even goes so far to say the mic doesn’t require the presence peaks or roll-offs typical of other dynamic mics, and virtually eliminates the need for EQ and processing. It’s also got a pneumatic shockmount system, offering the exceptional rejection of handling noise characteristic of Shure’s handheld mics. The KSM8 comes in two finishes (brushed nickel or black) and will be available as a handheld transmitter for use with Axient, UHF-R, ULX-D, and QLX-D systems. Jands: (02) 9582 0909 or AT 11


eMOTION LV1 SOFTWARE DIGITAL MIXING CONSOLE Waves is flexing its muscle in the live sound sector introducing the eMotion LV1 mixer, a digital mixing console that provides real-time audio mixing for FOH, monitor and broadcast engineers. Using the SoundGrid infrastructure for audio networking, it can be configured as a portable or stationary setup, and can connect to any SoundGrid-compatible I/O or server. Each channel on the eMotion LV1 has its own plug-in rack capable of running up to eight Waves and third-party plug-ins, so you can mix live with hundreds of instances of your favourite plug-

ins, all running inside the mixer itself. The mixer’s channel strip utilises the Waves eMo plugins: eMo D5 Dynamics, eMo F2 Filter and eMo Q4 Equaliser. eMotion LV1 comes in three configurations: 64, 32 or 16 stereo/mono input channels. The mixer can be controlled by hardware control surfaces and multitouch devices, ranging from four touch screens to a single laptop or tablet. Sound & Music: (03) 9555 8081 or

RADIAL’S DANTE DIs Radial Engineering has announced two exciting new additions to its respected line-up of audio accessories. The first is a Dante-enabled direct box called the DiNET DAN-TX with a 24bit/96k analogue to digital converter that allows for the input of instruments or line level sources into a networked audio system using the Dante protocol. Secondly, the DiNET DAN-RX acts as a 24bit/96k analogue endpoint that lets you output audio from

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a Dante network to stereo systems. It features two XLR outputs (left/right) and a level control that allows for connection to microphone inputs, up to +4dBu line level systems. A local 3.5mm headphone output is provided to test or monitor audio before connection to the PA system. Amber Technology: 1800 251 367 or

PRESONUS ULT SPEAKERS Presonus has released some loudspeakers of its own under its Worx Audio commercial division. It’s calling the ULT Series “the widest horizontal dispersion of any loudspeaker in their class with a focused vertical dispersion for an ultra-long throw.” The new line of active loudspeakers are designed for both mobile use and permanent installations. The series includes two full-range loudspeakers; the ULT12 and ULT15, and the ULT18 subwoofer. The ULT12 and ULT15 feature Presonus’ rotatable Pivot X110 horn and a 12-/15-inch, low frequency driver with a 2.5-inch voice coil. Both speakers are

bi-amped, driven by a 1300W peak power Class-D amplifier. Maximum SPL of the ULT12 is 132 dB SPL, and 136dB SPL for the ULT15. The ULT18 subwoofer is powered by a 2000W peak Class-D amp and uses a direct radiating, ported enclosure. A proprietary, 18-inch, low frequency transducer with a four-inch voice coil provides 7mm of driver travel before over-excursion, reportedly to push more air and provide more ‘thump.’ Link Audio: (03) 8373 4817 or

QSC E SERIES QSC’s new E Series range of passive loudspeakers have been designed with high-capacity continuous power ratings in mind to perform well with any professional power amplifier. When used as part of the complete E Series Entertainment System, the E Series takes advantage of advanced DSP settings for more flexibility, optimisation, and a higher level of performance. There are four speakers in the E Series lineup. The E10 has a 10-inch die cast frame woofer and a 2.5-inch voice coil. It’s ideal as a stage

monitor or main PA. The E12 is a trapezoidal twoway loudspeaker with a 12-inch woofer and threeinch voice coil. The E15, also with a three-inch voice coil, has a 15-inch 500W aluminium frame woofer, ideal for when high output and big low end are needed. The sub of the family is the single 18-inch E18SW, featuring a threaded M20 pole socket. Technical Audio Group: (02) 9519 0900 or

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BUILD YOUR OWN STRIP Not long after releasing the Virtual Microphone System, Slate Digital has introduced the second part of the equation, Virtual Preamp Collection, to all subscribers. VPC is designed to bring the authentic tone of two of the most famous analogue microphone preamplifiers to your DAW — the FG-73 models a famous British solid state pre, and the FG76 models a classic vintage tube pre from the ’60s.

Both modules feature ‘Virtual Drive’ knobs with compensated output, allowing you to saturate your sources through the emulated preamp circuitry while maintaining consistent levels. Those who are on the annual or monthly Everything Bundle can download Virtual Preamp Collection for free. Audio Chocolate: (03) 9813 5877 or

NOT-SO-NATIVE INSTRUMENTS Native Instruments introduced the Discovery Series: India — the latest in its line of Kontakt instruments exploring non-western musical cultures. The new library features a comprehensive selections of playable sampled percussion and melodic instruments and both north and south India with extensive performance control. Discovery Series: India features a modern interface that allows for ample sound customisation. An onboard

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groove player provides control and editing of rhythmic patterns, called ‘talas’, while 96 different scales called ‘ragas’ can be mapped to a keyboard in a number of ways to fit various harmonic contexts. The instrument features full integration with NI’s Komplete Kontrol S-Series keyboards. CMI Music & Audio: (03) 9315 2244 or

MARSHALL JMP 2203 GUITAR AMP PLUG-IN The Marshall JMP 2203 guitar amplifier plug-in was developed by Softube exclusively for UAD-2 and Apollo interfaces. It’s an emulation of the legendary 100W amplifier used by artists like Iron Maiden, Jeff Beck and Slayer. Introduced in 1975, the JMP 2203 is regarded by many as one of Marshall’s premier amp designs, known for its abundance

of dense crunch and power. The plug-in includes Universal Audio’s Unison technology, which means Apollo users’ guitar pickups will see the exact impedance load as if they’re plugged into a vintage Marshall amp. CMI Music & Audio: (03) 9315 2244 or

MOTU DP9 ENHANCED MOTU has been working hard on its flagship DAW Digital Performer 9, showing off a number of enhancements including better CPU efficiency for native plug-ins and virtual instruments, better latency performance for audio I/O and VIs, and new plug-ins for generating SMPTE time code and creating hardware inserts for outboard gear. These features will be supplied to DP users in two software updates over the next few months at no cost. The new Next-gen Pre-gen engine is the hero of DP9’s agility, transparently pre-rendering audio output from virtual instruments and plug-ins to

dramatically reduce their CPU load. Apparently in one bench test, Digital Performer was able to run 10 times the number of VIs with the enhanced Next-gen Pre-gen mode engaged, compared to realtime operation. DP’s host buffer has also been improved, now with half the latency than earlier versions. Using a MOTU 1248 Thunderbolt interface, DP achieved a roundtrip latency of 1.39ms with a 32 sample buffer size at 96kHz. Network Audio Solutions: (02) 9525 2088 or

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Hear Project Director for Constellation Systems at Meyer Sound, John Pellowe, talk about the Telstra CIC Theatre. Head to

Welcome to Telstra Theatre, nestled in its Customer Insight Centre (CIC) in downtown Sydney — the best-equipped theatre you’ve never heard of. The 300-seat Telstra Theatre isn’t a secret, yet its existence isn’t exactly common knowledge either. It’s had a few high-profile bookings (the 2015 ARIAs finalists announcement, for one) but it spends most of its time as a showcase of Telstra AV integration, packing the best of everything to blow away potential corporate clients. As you can see from the photo, it’s a stunning installation but what the photo doesn’t reveal is the most sophisticated audio rig you’ll find in any theatre venue in Australia; based on the Meyer Sound Constellation variable acoustic system. Constellation addresses a problem most performance venues suffer from: the need for multi-functionality. The issue is, multi-function venues often end up being ‘no-function’ or ‘multi-compromise’ — at least from an acoustical perspective. Aside for the likes of the thoroughbred opera houses or concert halls, most performing arts centres need to be as adept at hosting theatre, one-man comedy shows, cinema and rock ’n’ roll as it might be staging deb balls or travelling opera shows. The staging and lighting might be able to cope with the diverse roster, but often the acoustics don’t. A concert hall needs around 12 cubic metres per patron to achieve anything like an optimal acoustic for a symphony orchestra to bloom, while spoken word is best appreciated in a smaller/deader space with a commensurate ~five cubic metres per patron. The ‘happy medium’ acoustic of most

together with processing calibration ensure that feedback and phase cancellation are not issues. In fact, such is the sensitivity of the system that the theatre’s AV team mostly don’t bother with a lav or headset for on-stage presenters, rather, they’ll rely on the Constellation’s microphones to capture speech, which will be happily outputted to the system in a way that sounds utterly natural. Saying that, Anthony Lorraine, the Lead Theatre AV Specialist can virtually place theatre guests into a bathroom, the Grand Canyon or the acoustics of anywhere in between with the click of a mouse. Constellation’s CueStation control software has a Space Map feature that allows Anthony to map the theatre and draw in a custom panning curve — something impressively evident when he allowed the speakers and subs to stretch their legs by playing back a thunderstorm location recording that rippled dramatically around the room during AT’s tech tour. In audio terms, Constellation is the showstopper, it’s up there with the master conjurer making the Statue of Liberty disappear. Credit must go to acoustic consultants, Arup — initially led by Nathan Blum and then Ben Moore, and to the system designer (the audio luminary, Bob ‘Wrote the Book’ McCarthy) and the team from Meyer Sound. You have to hear it to believe it, and even when you do, you’re left shaking your head in amazement. — Christopher Holder

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performance venues is not happy at all. Meyer Constellation’s pitch is that it can electronically provide you with the perfect room acoustic best suited to the application. It’s a seriously tech-heavy solution and certainly not just a fancy surround sound system — there are far cheaper ways of setting up a hi-spec multi-channel audio system for flying sound around a room — Dolby Atmos or Barco Auro for a start. No, multichannel surround is more of a byproduct of the setup rather than the main game. To be clear, Constellation is not an artificial reverberation system, yet Constellation can change the apparent acoustics of the room, electronically. Here’s how: a matrix of dozens of Constellation microphones ‘sniff ’ or ‘sample’ the room in real time, feeding into a powerful Meyer D-Mitri DSP mainframe (the brain of Constellation). A newlygenerated soundfield, created in a natural manner by adding many finely-tuned delays to the input, is output through a multitude of critically-placed Meyer Sound loudspeakers. It’s enormously DSP-intensive but the results are truly spectacular. Constellation’s synthesised delays interact in a way that mimics real-world audio and become a natural component of the room’s resulting reverberant field, as treated audio blends with the room’s original audio. It’s not simply about hearing the performer in the right environment, you as the patron share that same space — rustle your packet of Jaffas, or sneeze, and you hear yourself in the same ‘concert hall’ or ‘cinema’. The microphone and speaker positions,

Meyer Sound: (07) 3267 7800 or Telstra Theatre: AT 17


Keith Richards almost quit music, but now the Stones’ guitarist is under its influence again. Story: Paul Tingen

Album: Crosseyed Heart Artist: Keith Richards AT 18

“Keith is very charismatic, very smart and a very sharp guy. He reads a lot and is an excellent observer. He’s also very down to earth and humble; there are no airs.” Dave O’Donnell’s — who was the engineer and mixer of Keith Richards’ third solo album, Crosseyed Heart — generous depiction of Richards are in line with his current status as venerated elder-statesman-of-rock. There was a time when wider perceptions were different — all of the ’70s, in fact, when almost all of the rock ’n’ roll community was forecasting his imminent drug-induced demise. Fortunately, its Richards love of music that has given him the desire stay upright, and to endure the rigours of life on the road with the Rolling Stones. A recent film documentary about Richards, Under The Influence, gets the story straight from the horse’s mouth. The film is essentially an album promo, but the intersecting file footage validates the claims he was indeed ‘there’ at turning points in American Blues, Country and Jamaican Reggae. And loving it no less. If you’re to believe this documentary, it’s this same soul-lifting joy and exuberance divined from new musical discoveries that underpins Crosseyed Heart. JUST DEMOS, CROSS MY HEART

Twenty-three years, the entire lifetime of some artists, separates Crosseyed Heart from Richards’ last solo album, Main Offender. While the gap between that and his solo debut, Talk Is Cheap, was only four years. All three albums feature a collection of Richards’ musician friends called the X-Pensive Winos — drummer Steve Jordan, guitarist Waddy Wachtel, keyboardist Ivan Neville and backing vocalist Sarah Das. This album also saw guest performances by keyboardist Spooner Oldham, pedal steel-player Larry Campbell, and vocalists Aaron Neville and Norah Jones. The musical palette is rich with Robert Johnson-influenced acoustic blues (the title track), Stones-like rockers (Trouble, Heartstopper), reggae (a cover of Gregory Isaac’s Long Overdue), folk (Ledbetter’s Goodnight Irene) and several gorgeous ballads (Robbed Blind, Suspicious and Illusion). Even at 15 tracks and a length of nearly an hour, there’s relatively little filler. Richards’ first two solo efforts received mixed reviews and middling commercial success. Crosseyed Heart, by contrast, was, at the time of writing, set to crack the Top 10 in the Australian, US and UK album charts. The reviews have given it ‘best yet’ status, with many commenting on the “relaxed” and “spontaneous” atmosphere of Crosseyed Heart’s “ragged set of no-nonsense blues tracks,” which have a “satisfyingly gritty texture, more stripped back than a Stones album.” Crosseyed Heart sounds like Richards and the Winos had a rollicking good time. Which is partly due to the way Jordan — who co-produced and cowrote — and Richards approached the project. In Under The Influence, Jordan explained he contacted Richards after hearing an interview where the rockstar suggested his own retirement. It didn’t impress Jordan much, so he prescribed the studio

as a tonic. Just the two of them throwing down “a few ideas… not an album.” It’s a good tonic, reckons O’Donnell: “Keith loves being in the studio. He’s made a lot of records and understands the entire process but still enjoys being there while we’re working, even when we’re fiddling around with something technical. “In early 2011, Keith and Steve had been improvising together and co-writing for a while at One East Recording in Manhattan. In March of that year Steve invited me to come to Germano Studios, also in Manhattan, where most of the album was eventually recorded. They said they weren’t planning to make an album, but just wanted to record some material they’d been working on. In the beginning we worked maybe two days a week, and got a song recorded each day. It was all very relaxed. You never know what people are like until you actually work with them, and Keith has incredible feel. His playing is purely natural; there’s no ‘work’ involved.”

On the opening track of the album, Richards immediately throws down the gauntlet to anyone who assumes he’s no longer the player he once was


In addition to the quality of the songs, several elements contribute to the success of Crosseyed Heart: the intelligent arrangements and playing of the guest performers, Richards’ surprisingly strong vocals, his formidable guitar playing, which is all over the album, and the warm, deep, inviting, 1970s-yet-modern sound image. On the opening track of the album, Richards immediately throws down the gauntlet to anyone who assumes he’s no longer the player he once was. Crosseyed Heart features a brilliantly executed acoustic solo guitar, played in a 1930s blues style — tightly swinging and showcasing some impressive finger-dexterity. O’Donnell elaborated: “He came in one day with an old 1940s Martin acoustic and said to Steve: ‘there’s this thing I want to play for you’. Pierre de Beauport, who is Keith’s guitar tech, and I both know that if Keith wants to play, you’d better be ready to record. If I or our assistant Kenta Yonesaka could not get out in time to place the mics, Pierre would do it. Keith took out his guitar, played and sang, and finished by saying, ‘that’s all I got’. That, literally, was it. It was just one take, and everybody loved it. It’s great blues playing that really showcases how stunningly good he is. THE MICS WE USED ON KEITH’S GUITAR FOR THAT TRACK WERE A TELEFUNKEN M260 CLOSE UP, PLUS A LARGE DIAPHRAGM A BIT FURTHER AWAY, PROBABLY A NEUMANN TUBE 47, and Keith sang into a Shure SM7, which is unidirectional, so didn’t pick up a lot of guitar (see Recording Keith’s Acoustic Guitars for more details). “People sometimes talk about Keith being ‘loose and sloppy’, but he is an amazing guitar player! He’s like an old blues guy. The feeling just emanates, and he was always in tune. When he picks up an acoustic guitar, he’s all over that thing! “When playing electric he uses very few pedals, it’s mostly just straight into the amp. THAT’S REALLY HOW YOU GET THE BEST SOUNDS; A GREAT GUITAR INTO A NICE AMP, NOT PLAYING TOO LOUDLY. YOU GET A

This is also true with acoustic guitars. You hear the dynamics; Keith MUCH BETTER TONE THAT WAY.

has that down, and James Taylor is the same. They can play very softly and get a great tone, then suddenly hit hard.” JUST THE TWO OF US

Most of the remaining time at Germano was spent with Richards playing guitar and singing guide vocals, with only Steve Jordan on drums. As a rule, other instruments, and lead and backing vocals, were overdubbed later on. “The room is maybe 10m long,” recalled O’Donnell. “Steve’s drum kit was set up in the back corner of the studio live room, facing me. Keith mostly played electric guitar facing Steve with his back to the control room. The guitar was at a good volume, but not so loud that it would drown out the drums. We weren’t looking for isolation anyway. Keith used headphones if he wanted to hear himself singing, and Steve had a live monitor wedge so he could hear Keith’s singing and more guitar if needed. “The songs came into being in different ways. Keith had written some of them before recording, like Robbed Blind, and others he improvised and cowrote with Steve. They played almost all the basic tracks together, so it has a live feel. After a few takes they’d come into the control room to listen, then either redo it or change sounds and play it again. If a song was taking longer to record, they’d revisit it another day. We faded out or edited down some songs that were played for longer than needed, but there was no editing between different takes.” O’Donnell recorded everything to 24-track analogue tape — running Pro Tools at 24-bit/96k as a safety — which he said, contributed significantly to the warm sound of the album: “WE TRACKED TO A STUDER A827, RUNNING AT 15IPS, BECAUSE WE WANTED THE ‘LOW END THAT PROVIDES’. While there

was no intention to make it sound like the Stones, with Keith Richards in the room you can’t help but be in that vein. Keith and I didn’t really discuss AT 19


“We recorded his acoustic guitar for Crosseyed Heart with a new Telefunken M260, which Germano Studios had recently acquired. I’m sceptical of new mics, because the older ones have been around forever and you know what they do, but this one sounded fantastic. I loved it right away, and later used it with James Taylor. The other acoustic guitar mic was a large diaphragm, a tube Neumann U47 or perhaps a Telefunken ELAM 251, and Keith sang into a Shure SM7. All the mics went through the studio’s AMS Neve 1084 mic pres, and then to tape. I placed the M260 between the sound hole and where the neck meets the body, slightly angled towards him, and the other mic somewhere within 2-3 feet of the body of the guitar. “Acoustic guitars are tricky to record, because the player can move around a lot, but I do like the sound between the sound hole and the 14th fret, because it gives you a lot of definition. I may have it just 15cm from the guitar, but if it gets too boomy, I back off the mic a bit. Pointing the mic at the neck has never worked for me. I decide on the placement of the other microphone by listening in the room, and also by what happens to the phase. Things can be out of phase and sound bad, and you move the mic a bit, and suddenly it may sound good. “The second mic can be cardioid or omni, depending on the room. When recording someone whose fingerpicking, like Keith did in this case, I’ll track it with an 1176 compressor, just in case some of the snaps are too hot on the close mic. I’ll slightly pan the two mics, and in the mix later on we added a touch of reverb. I liked it dry, but Keith wanted a little more space around it, so we added some Bricasti, or perhaps the old EMT250.”

People sometimes talk about Keith being ‘loose and sloppy’, but he is an amazing guitar player! He’s like an old blues guy. The feeling just emanates, and he was always in tune.

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ahead of time how he wanted things to sound, but I know he loves analogue. So does Steve, to a huge degree. We also always ran a two-track tape recorder for tape-slap throughout tracking and mixing. Whenever Keith came in to listen, he would always see one or two tape recorders running, which he clearly enjoyed! “Tape definitely adds character. For this music it was part of the sound and contributed to creating the right vibe. We would record the kick and snare — and sometimes the drums as a whole — quite hot to get some tape compression, but the guitars didn’t really need that. Steve and Keith would be happy to work 100% in analogue, and so would I, but digital does provide a number of advantages, especially when you’re not specifically trying to make a record but are ‘just putting down ideas’. With tape you always have the panic of, ‘oh, they’re doing a great take, is the tape going to run out?’ and then almost hoping the musicians either stop, or someone makes a mistake! With Pro Tools you can record for as long as you want. Keith overdubbed his bass

to tape, but after that we generally transferred everything to Pro Tools to open up more tracks for experimentation during overdubbing.” FIRST NAME BASSIST

Once the guitar and drums were laid down, Richards would overdub bass guitar. Then sometimes on, through an impressive array of instruments like more acoustic guitar, piano, Wurlitzer, Farfisa organ, electric sitar, and tiple, plus his lead and backing vocals. “Steve wanted to get as much of Keith on the record as possible, and was well aware he’d played bass on some of the greatest Stones tunes,” related O’Donnell. “He plays melodically and rhythmically, and reacts to his own guitar playing, so he comes up with original stuff another bass player wouldn’t necessarily play. “He played his bass overdubs in the control room through an Avalon Tube DI, and also through the studio’s Ampeg bass amp. I generally like a Neumann FET 47 on the bass cabinet, but the mics changed. We could have used an EV RE20, or a


O’Donnell: “As with any instrument, the sound really comes from the player’s hands. If you gave Keith’s guitar to another player, it’d sound completely different. Of course, the guitar and the amp do make a difference, and in general Pierre [de Beauport] would hand Keith a guitar, and he played almost everything through a Fender Champ, certainly all his rhythm parts. “The guitar most often used for the rhythm parts was his famous ‘Micawber’ 1953 Fender Telecaster, which is in open G tuning and only has five strings — the lowest string having been removed. That’s his main guitar for riffs. For solos and also on the acoustic he tended to be in regular tuning. “I did what I normally do, which is to have a Shure SM57 close up on the cabinet, and sometimes I also had a Sennheiser MD421. Because we were only recording two days a week, we’d have to tear down and set up regularly, so we might have changed the mics. But generally it was the 57, and often a 421 blended in. For some songs we had a Royer 121 behind the cabinet for more low end, and a Beyerdynamic M160 for some ambience. As for positioning, I will always walk around and listen. The close mic is rarely straight onto the speaker cone, but always off-centre and angled slightly. Sometimes I’ll swap the 57 for another one, as they all sound slightly different. The mic pres were the AMS Neve 1084s at Germano Studios. It’s possible we sometimes used an 1176 on the way in as well, but that would have been minimal. We also recorded a DI as a safety, with the idea that we could re-amp later, if necessary, but we never had to. Obviously the sound was right, because it inspired him to play.


“Waddy Wachtel played guitar on half a dozen tracks through the Watkins amp. We had a similar recording setup for him; an Shure SM57 and Sennheiser 421, a Neumann 87, and an ambient tube U47. We removed the top and front of the studio’s upright piano, and recorded Keith playing it with a pair of Neumann U87s at the back and a couple of AKG 414s at the front. “Steve would often change the sound of his drums for each song. We usually had a 421 and FET 47 on the kick, the snare would have an SM57 top and bottom, the hi-hat would be a Neumann KM84 or KM184, and we’d have 421s on the toms. Steve had a second kit set up in the other corner, which would have even more sparse miking. Most of the drum sound came from the overheads, which could be Coles ribbons, Telefunken CineMics, or Neumann U67s. Keith and Steve

specifically did not want a multi/close-mic sound for the drums. The idea was to have a more open, raucous sound for the album. I’d say we weren’t going for a lo-fi or hi-fi sound, but what I’d call a mid-fi sound; more like ’70s rock ’n’ roll. “Everything, including the drums, went through the AMS Neve mic preamps — either 1084s or 1081s. If those were not available we’d use the Chandler TG MkII.” MIXING STUDIOS

Beyond Wachtel and Richards’ overdubs, Paul Nowinski played viola and gamba, as well as double bass on the ballad Robbed Blind, Pino Palladino played electric bass on Illusion, for which Norah Jones co-wrote the lyrics and overdubbed her vocals. O’Donnell recorded her with a Neumann U47, the same mic he tended to use on Richards’ vocal overdubs, in addition to “occasionally an RCA 44 or a newer Telefunken”. All overdubs were recorded at Germano Studios, with the exception of horns on Lover’s Plea and

“With guitar overdubs we spent more time looking for different sounds, and sometimes changed the mics — it could be an SM57, 421, Neumann FET 47 or AKG C12, depending on the sound Steve and I were after. Keith still almost always played through the Champ, though he occasionally used a Watkins amp. Pierre supplied him with different guitars, and occasionally suggested a pedal to get a specific tone. “You can get a certain sound from lead guitars by turning them up loud, so we tended to use more ambient mics to get a bigger, more exciting sound. Lead guitars tend to come and go in the song, so they can take up more space in the track. “Keith came up with many interesting guitar parts, which were all very feel-based. He’d come up with ideas, do a couple of takes, and might refine his idea and play it again. We comped some of the overdubs, but it’d be like the first half of one take and the second half of another take. It was all about capturing the spirit of things, never about being note-perfect.”

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Steve wanted to get as much of Keith on the record as possible, and was well aware he’d played bass on some of the greatest Stones tunes.

Charles Hodges’ Hammond organ and piano overdubs, which took place at Royal Recording Studios in Memphis, Tennessee. O’Donnell said it was only when almost everything was recorded that the ‘just layin’ tracks down’ narrative was overtaken by record making: “We had done rough mixes as we were recording, but hadn’t spent a lot of time on them. At some point the decision was made to release the material as an album, which meant mixing it properly. The mix phase was more conventional in that we would go in for four to five days a week. Keith and Steve were involved the entire time. Keith wanted to be there, he loves that kind of teamwork. Because many of the roughs were very good, we just finished them off at Germano, going from Pro Tools through the SSL Duality. After that we took some songs to Brooklyn Recording to use their Neve 8068 desk and incredible array of outboard. “We only used outboard during the mix, no plugins. It was all old stuff, like Teletronix LA2As, UA 1176s, EMT plates, and of course the tape machine slap echo. WHAT DID I DO FOR GUITAR TREATMENTS? JUST PUSH THE FADER UP! SERIOUSLY, THERE’S NO BIG SECRET OR MAGIC BUTTON. The sound really comes from the

hands, the guitar, and the amp, and then of course AT 22

the mic and the mic pre. Mixing mostly consisted of balancing, panning, and EQ’ing on the Neve console if we felt it needed a bit more definition. There were quite a few guitar overdubs; you can listen to the album purely for the guitar parts. “Steve constantly came up with great ideas, but mostly we had achieved the sounds we wanted during the recordings. Some of the overdubbed guitar sounds might have been treated a bit more. If we wanted a compressed sound, we’d use an LA2A, 1176 or ELI Distressor, which is one of the newer pieces of gear that is excellent. We also used the tape slap quite a bit during the mix. PANNING IS HUGE FOR ME; I LIKE LEFT-CENTRE-RIGHT FOR GUITARS. THE STUFF IN BETWEEN IS MORE FOR AMBIENT THINGS.

“We had the Vertigo VSC-2 quad compressor over the two-mix on many of the songs, and we printed to both ½-inch tape and Pro Tools and brought them both to mastering. Greg Calbi mastered it, and his first question is always: “How loud do you want it?” We said, “We want it musical!” Which Greg liked. We wanted Keith’s record to sound modern, but not overly bright or compressed. Any distortion that’s there was intentional, created at the time we recorded! For me it was all a matter of capturing the spirit.”


Dave O’Donnell took his first studio steps in 1984 at the legendary Power Station Studios in New York (now Avatar Studios), as a runner. He quickly worked his way up to engineer, working with legendary producers like Russ Titelman, Phil Ramone, and Neil Dorfsman. O’Donnell went independent in the early ’90s, and has amassed an impressive list of credits, including the likes of Eric Clapton, Keith Richards, John Mayer, Lyle Lovett, The Bee Gees, Milton Nascimento, Rod Stewart, Joss Stone, Morrissey, Tina Turner, James Taylor, and Ray Charles, and has been nominated for three Grammy Awards, winning one. His multi-album collaboration with James Taylor has been particularly fruitful. Among the several Taylor albums he’s worked on, O’Donnell engineered and mixed the guitarist’s October Road and earlier this year he engineered, mixed, and produced Taylor’s first-ever American No. 1 album, Before This World. In 2013 O’Donnell mixed a track featuring Clapton and Richards on the Crossroads Guitar Festival 2013 album. O’Donnell has a project studio at his home in upstate New York, called Studio D (top), with a Yamaha DM2000 desk, a Pro Tools HD system, ProAc Studio 100 and Griffin G2B monitors, and several racks of outboard gear.

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We follow Welcome to 1979’s Chris Mara through the edge of your seat process of direct-to-disc recording and find out what it truly takes to ‘cut’ a record. Story: Mark Davie

AT 24

There’s an ingredient lacking in the Raw Food movement — immediacy. If I’m going to eat uncooked superfood spheres followed by a serve of matcha hemp chia pudding, I bloody want it now! You know what good raw food is? Grabbing a carrot out of the fridge and chomping on it like Bugs Bunny. That’s ‘what’s up’ — raw, untouched, pure unadulterated carrot, direct to my mouth without mashing it into some tasteless tahinitempered puree. Chris Mara thrives on immediacy. He’s built a business out of it; well, several business actually. There’s his studio, Welcome To 1979, a purely tapebased Nashville studio he built eight years ago to keep track counts low and sonic austerity at a high. Though, he admits, the decision to not secure a Pro Tools rig at the time was as much about business as sonics. “I had a financial choice to make,” reasoned Mara. “Getting a 24 I/O Pro Tools rig was about $40,000. The place I wanted to rent was about 8000 square feet. I already had a two-inch 24-track, my console and a bunch of outboard gear. Why spend 40 grand to be just like everybody else and then have that payment? I could not get Pro Tools, and that would be my thing.” Welcome to 1979 drew notoriety for its ‘hard line’ stance, and other engineers started calling up to ask where Mara had his machine restored. “I restored it,” he’d explain to them, which spawned another cottage business, Mara Machines, which can supply any flavour of completely restored MCI tape machine — from a ¼-inch two-track to a two-inch 24-track — for an affordable price. “In the first year we did one every other month and it helped fund the studio because it was extra money. EVERY YEAR IT’S BEEN DOUBLING. THEY GO EVERYWHERE — GREECE, MEXICO, CANADA, VIETNAM — LAST YEAR WE DID 50. WE’RE BY FAR THE LARGEST TAPE MACHINE RESTORATION COMPANY IN THE WORLD. I’ve got a warehouse with

about 40 of them in there right now. I bought two today and two on Monday.” Eventually, Mara relented and installed a Pro Tools rig in the studio because “we do so many transfers and I do a lot of mixing in the box. We track to tape, the band goes on the road, I transfer it to Pro Tools and send them mixes.” Having a digital rig also provides a playback solution for his most immediate recording solution; direct-to-disc recording. CUTTING, A NEW HIGH WAY

Literally ‘cutting records’ from a lacquer as the artist is performing live is as much science as art. Welcome to 1979 only got into cutting its own lacquer masters about three years ago. Because many of Mara’s clients came to him specifically to record on tape, they were also more likely the type to take that next step down nostalgia highway and press vinyl. “I started sending a lot of projects off for other people to press and some sounded really good while others sounded really bad,” said Mara. “I wasn’t sure why because I didn’t know anything. So I called a mastering engineer to ask what was going on. He said I should get into cutting my own masters and offered to teach me how. I bought a lathe, had it restored and almost immediately

started cutting a lot of records.” But it didn’t stop there. Mara: “The guy that restored it for us said ‘You’ve got to cut direct to this sometime, since you have a studio connected to a lathe, and they’re in the same building. That never happens.’ Usually the lathes are in mastering studios and there’s no way you could physically do it.” So over the last year, Mara has started cutting the odd project directly to a master lacquer, the sixth and most recent was Josh Hoyer & The Soul Colossal’s Cooked Raw — fittingly, the record has six tracks from the six-piece band. Hoyer and his ensemble tour hard — they’ve put 75,000 miles on their van in the last two years — and a lot of their fans appreciate the raw, uncooked live version. “They tell us they love our studio albums but it doesn’t capture what we do live, which is where we sound the best and our energy is. After all, that’s what soul music is supposed to do, connect with people,” said Hoyer. “So I said, ‘Let’s just give this a shot, go direct-to-disc, and capture that energy.’” There were obviously things that were out of the question: backup vocals, certain layering techniques… “But I like how it came out,” said Hoyer, who enjoyed the process. “It is what it is, like the jazz recordings I’ve listened to for so long. There were definitely lots of upsides. We took some chances. The leads — whether, trombone, sax, guitar or whatever — can play off of each other better. You’ve got that immediacy and interplay that helps bring the groove together. You capture much more energy than when you’re just playing one instrument at a time. We were all able to watch each other. I mean, it’s like playing a show. We’re probably going to do another one down the line.” While there wasn’t room for any overdubs, it didn’t stop Mara and Hoyer getting a little inventive with the setup. Hoyer usually plays a combo keyboard, switching between B3 and Wurly patches on the fly. The day before the session, the keyboard got dropped after a gig… perfect timing. Luckily Welcome to 1979 has a healthy collection of vintage instruments, including both a Hammond B3 and Wurly in great condition. “We ended up putting the B3 in his vocal booth with the Wurly off to his left,” explained Mara. “IT WAS THE LAST THING WE DISCOVERED. I HAD TO PUT UP TWO VOCAL MICS (ONE OVER EACH KEYBOARD), BUT I DIDN’T HAVE A MATCHING PAIR. SO I SET UP A NEUMANN U47 AND MIKTEK CV4, EQ’D THEM TO SOUND THE SAME, AND BUSSED THEM TO THE SAME COMPRESSORS — AN AVALON 737 WITH A DBX 160 SET UP BEHIND IT ACTING MORE AS A LIMITER.” For the whole

session Mara watched Hoyer like a hawk, switching faders as he jumped between the two mics. “He would do it mid-sentence!” exclaimed Mara, forcing him to participate in the session as ardently as the band. “He was kind of in the room playing along with us,” said Hoyer. Along with Hoyer’s double keyboard and vocal combo, the Soul Colossal has a drummer, bassist and guitarist, and two horn players. Mara put the rhythm section in the live room, taking a direct line from the bass and combining it with the amp signal. The guitarist’s amp was isolated, as were Hoyer’s B3 and Wurly amplifiers. The studio also

has a large entryway between the live room and vocal booth, with windows looking into each. It was a perfect position for the horn players and meant everyone could see each other. Once the setup is locked in, the band will record one side at a time. They’ll go in, perform a four song set leaving a couple of seconds between each song, then cut another four songs for the other side. Mara reckons four is a good balance between getting the most out of a side while limiting howlers. In this case, it was three a side. The song order is just down to common sense; what feels right, and eliminating any unnecessary changeovers like capos and tunings that mean you’ll be swapping guitars every song. Having one guitar player also cuts down on tuning issues over four songs.

It’s sonically one of the best ways you can record because it’s literally microphone, console, lathe — you hit a note and it’s on the master lacquer in a fraction of a second


Mara wasn’t coming to the process cold. Over the years, he’s done a dozen or so live to ¼-inch twotrack records for the Upstairs at United project. In a room above the United Record Plant, Mara would cart in an all-analogue setup of a 16-channel console, some outboard, and mics to capture performances live to tape. The lineup of musicians was diverse, from Jack White-offsider Brendan Benson to American indie band Cults, British piano rockers Keane and Australia’s urban cowboy, Henry Wagons. It was great practise before taking the next step towards tracking direct-to-disc, which adds a level of live complexity at the cutting head. For a quick education Mara and mastering/ cutting engineer Cameron Henry turned to mastering engineer Hank Williams for a masterclass on how to operate the lathe. “He turned us into experts in a short amount of time,” said Mara. “Because he’s so knowledgeable.” AT: What are the limitations of the medium? CM: “We can pretty much put anything on it. Because it’s a physical medium, it has physical limitations — mainly on time and volume. The louder the record the shorter it is and vice versa. YOU WANT A RECORD TO BE LOUD BECAUSE VINYL RECORDS HAVE A NOISE FLOOR AND YOU WANT TO BE ABOVE THAT AS MUCH AS POSSIBLE. AT 25

“Playing back is where the issue always is. The stylus is going left and right for mono and up and down for stereo. It’s moving around, and we have to rein it in enough so the music and energy doesn’t change but everyone can play it back on their turntable. “On a typical cut, you’d send us your music and we would audition it; listen to it, often do a test cut and listen to that, then do a master cut. This is all on the fly. We play it a little bit safer than normal because we don’t want to capture a perfect performance then find out it doesn’t play back on a turntable because the kick drum’s too loud. That said, you take a master to the plant and there’s two more processes that need to take place. They could still totally screw the pooch on it.’ AT: Sounds like you need a few hands just to make it work. CM: “On a direct-to-disc session it gets pretty complicated because I’m mixing in the control room with my assistant, and Cameron’s in his mastering suite with his mastering assistant. Our assistants are on the phone to each other, because you don’t just say, ‘Rolling’. He has to drop the thing down and wait for the lead in, then say ‘Go’ before I can tell the band to play. It’s not instantaneous, and he could have a problem that forces us to stop. There’s a lot of choreography there, but it’s pretty fun.”

AT: Once everyone’s rolling, what happens next? CM: “Cameron’s highly skilled. When you tell guys that cut a lot of records that we do direct-todisc, they kind of go, ‘Damn!’ On a standard cut one of the main things our lathe and models like it can do, is change the distance between the grooves based on how loud the music gets. It means you don’t waste space on the record, and can have a louder record. “On a direct-to-disc you can’t use that system, because it’s live. Cameron is listening, watching and changing the spacing manually. IF IT GETS QUIET, HE’S LOOKING AT THE MICROSCOPE TO SEE HOW CLOSE THE GROOVES ARE AND AS IT GETS LOUDER HE STARTS TO WIND IT OUT SO WE CAN STILL HAVE A GOOD, LOUD RECORD. IT’S CALLED ‘PACKING THE GROOVES IN TIGHT’. It’s pretty

impressive to see one under the microscope that’s done really well.”

I’m not trying to demean anyone with Pro Tools skill sets, I have them too. But it really is a challenge to do things direct-to-disc. Not many engineers today have ever done it


While there aren’t any edits done to the performance, the band does get multiple shots at recording a side. But when it comes time to choose which performance was best, there’s a problem: You can’t listen back to the master lacquer because it will degrade it. Getting around that issue has a two-part solution. RIGHT AT THE BEGINNING OF THE SESSION, THE BAND RECORDS A REFERENCE LACQUER AND LISTENS

MARA MACHINES RESTORATIONS AT: Why do you only restore MCI machines? Chris Mara: In Nashville in ’95, you made records on tape machines. As an assistant and then as an engineer I got familiar with how they sounded. Also, they break and it seemed like the MCIs, even before I knew how to restore them, were very approachable from a technical standpoint. They were made to be fixed and sounded the best to me. Studer and Otari’s definition of sonic quality is to be neutral, which is valid and hard to do, like a Grace Design mic preamp. But that’s not what I want out of a tape machine and certainly not what people want now. You don’t buy a tube microphone for its frequency response — you want curves, valleys and high-end roll-off — that’s what the MCI does. They’re made here in America which makes parts a lot easier. I don’t know exact numbers, but MCI made a shit-ton of them. They designed the circuits around parts that are still available. They didn’t say, AT 26

“Hey, go build this for me.” They said, “What do you have? Let me design around it.” The power switch to turn it on is $4 at the hardware store. You can’t find the power switch for a Studer now. AT: What do you look for to make sure a tape machine’s not beyond saving? CM: I just make sure the machine is complete. Everything can be fixed, but it’s harder to buy things for it. I’ve got a lot of pieces and parts lying around but I try to make sure every machine that comes in goes out. I don’t want to have a hundred carcasses to walk by every day. AT: What typically goes wrong with them? CM: There are two main parts to the machine — the transport that pulls the tape and the audio. It’s like getting a good drive in golf and three-putting. If the transport’s easy then the audio is gonna be a bitch. They all come out costing roughly the same amount

of money and time, just spent in different areas. Over the years I’ve developed a three-page checklist of things we do before we even turn them on. We tear them apart, replace all kinds of parts that may go wrong, turn them back on and then it’s only a couple of weeks of fixing things because you’ve eliminated a lot of potential issues. AT: What’s a fully-functioning, restored two-inch 24-track worth? CM: Surprisingly little. I’m selling them for US$8250. It should go up a bit. I don’t raise the price because I make a bit of money on that, and I like who’s buying them. I don’t really want to supply super, high-end assholes. The guys who are buying them are into it. It’s like buying a high-end microphone. I’m not saying eight grand isn’t a lot of money, but it’s not $25,000. It’s achievable if you’re serious about it and it does a lot more. It’s bigger. I try to quote people a price per pound!

MICS & MORE MCI It’s MCI through and through at Welcome to 1979. Mara used around 24 channels on his late ’70s MCI 428 console for the session. Here’s what he used to mic up the instruments. Mara: “I used a couple of mics on the kick drum, an AKG D112 and a Neumann U87. I usually use a subkick, but I don’t on direct-to-disc because it’s too risky having that much low end. I put a Shure SM57 on snare, and put up a couple of tom mics. I used an RCA 44 ribbon for a mono overhead. Stereo imaging just for the sake of it causes you to waste space on a record because the needle has to go down, which means the grooves have to go wider. Tighter drums means the record can be louder. “I used another RCA 44 on the horns. The bass amp was a cool mid-’60s Ampeg B15 flip top. The guitar player had a Fender Twin and we used a 57 and an AKG C12 right next to each other and right up on the grille. I run those to one compressor and one track. I never liked putting them on separate tracks because you never get the blend right and they don’t hit the same compressor. The initial blend is the EQ because one’s softer than the other. I just get that set and make sure they’re very phase coherent. When the guitar player stands there with noise coming out his amp, if I don’t hear noise then I’m good. I like using a dbx 160 compressor for guitar, because when you lean on it, it gets beefier not dark. We had a 57 on his Wurly amp and a couple of Neumann KM84s and a D112 on the B3 Leslie.”


If everything sounds good to them, they sign off on it, and recording proper gets underway. During the final performances, the outputs on the lathe are run into the Pro Tools rig, though Mara is quick to stress, that there’s nothing between the console and lathe. A duplicate of what’s gone down is recorded into Pro Tools so the band can listen through all their six takes and decide which is best without messing up the lacquer. “It’s fun to watch bands listen to them,” said Mara. “Because they’re not just listening to their part. They’re listening to the whole thing. It’s about weighing up the mistakes against the energy.” OKAY TO PLAY IT BACK.


From a recording engineer’s standpoint, Mara likes the idea of challenging himself and his skill sets with more analogue productions. “I’m not trying to demean anyone with Pro Tools skill sets,” said Mara. “I have them too. But it really is a challenge to do things direct-to-disc. Not many engineers today have ever done it.” During the performance Mara would “ride guitar solos and the occasional snare fill. If it got to a down section on drums I would bring the drum room mics up a little bit to fill some space. They have such good dynamics that they kind of mix themselves. Even an overall static mix had a lot of dynamics in it. I would also grab more reverb in the keyboard and guitar solos as well.

AT 27


muting and unmuting his two vocals mics, but it sounded too obvious, so I faded them instead. And at the end of the side I’m doing a fade.” At the end of the chain, Mara has a stereo compressor and a limiter, just to round it off a bit. Henry also has a compressor and limiter in the lathe room, with a multi-band EQ so he can shape the signal. “I don’t have to think much about it,” said Mara. “I can just make a good sounding mix and rein it in a little bit. We don’t want to squeeze it because the more dynamic it is, the louder the vinyl record will be. “If you sent me a torpedo, the record will be quiet because it can’t spread the grooves out and slam them in. They’re all just spaced the same. We try to keep the dynamic range but squeeze it a little because we’re not sure who’s going to jump in. Hoyer’s got horns and he’s very dynamic vocally, so it’s a mixture of leaving it alone, then if it starts running away we have processors that will clamp it down.” MAKING THE CUT

So why go through all the pain of cutting a record directly to lacquer? Mara reckons there’s nothing AT 28

quite like it. And he’s right. Mara: “It’s the trifecta. It’s sonically one of the best ways you can record because it’s literally microphone, console, lathe — you hit a note and it’s on the master lacquer in a fraction of a second. It’s fast because you come in for a weekend, or three or four days, and your record is done — it’s tracked, there’s no mixing, no overdub, no mastering, nothing. AND IT’S AFFORDABLE BECAUSE WHEN YOU WALK OUT OF THE STUDIO, YOU HAVE YOUR MASTER AND YOU’RE ALREADY ONE-THIRD OF THE WAY INTO VINYL PRODUCTION. RARELY ARE THINGS FAST, INEXPENSIVE AND GOOD.

“Another nice thing is that no one’s doing it. It has a story built in, which you kind of need these days to stand out. People are over, ‘Hey, I recorded in a cabin in the woods.’ Or, ‘I recorded at home with a laptop.’ Everybody’s f**king doing that.” Mara’s adamant that any act that plays their instruments could handle the pressure. After all, he says, “It’s basically like playing a four-song set twice, which bands do all the time. They go out and play four songs in a row without major screw-ups. It’s not impossible.”

You capture much better energy than when you’re just playing one instrument at a time. I mean, it’s like playing a show

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Kurt Vile hops between studios up the West Coast to avoid the ticking clock and winds up recording the next Loser with Rob Schnapf. Story: Mark Davie

Artist: Kurt Vile Album: B’lieve I’m Goin Down AT 30

“It wasn’t trying to be like the Foo Fighters record.” Kurt Vile’s bandmate Rob Laakso assured me no cable TV money and syndication rights prompted their 12-stop nomadic recording trek for his latest album B’lieve I’m Goin Down; just a journeyman trying to reassess his creative process. It’s not uncommon for Kurt to drop in at multiple studios over the course of making an album. The record before last, Smoke Ring for My Halo, started out in a couple, he said, and blossomed from there. But this time there was a bit more purpose behind his choices. Halo was more of an East Coast record; this time, he wanted to stick mostly to the West Coast so he could play with Stella Mozgawa, the drummer for Warpaint, and Farmer Dave, slide guitar player and roaster of hot nuts. “You can tap into other worlds and atmospheres as opposed to flying everyone to you,” explained Vile. “Which feels more contrived.” Those feelings surrounding the process are important for Vile, whose music is largely introspective. There was another he’d carried for a while, but peculiarly for a songwriter, never been able to fully articulate. During the making of the five records before B’lieve I’m Goin Down there were occasions he felt the process had robbed him of the time to do particular songs justice. He wanted to see if he could correct that imbalance. “One example I have is my song Peeping Tomboy,” offered Vile. “I was really feeling it when I wrote that. My wife was away and I knew I was just about to be fired from my job, like psychic or something. I had just signed to Matador, so it didn’t matter, but life was so uncertain; my wife was so uncertain. “I would play the song live and really get into it. My idea was to try to sound like I’d just written the song all alone in my house in the middle of the night. Just wait to capture that certain vibe. But when it was finally time to lay it down, I don’t think my performance was that good but it’s the best one there was at the time. I think part of it was just being in this big studio and I was nervous with all these nice mics around and it really moved fast.” His last two records, Wakin’ on a Pretty Daze and Smoke Ring for My Halo, were recorded with John Agnello, who Vile says was “totally on the team, but you still had to talk about schedules and pick the studio far in advance.” This time, Vile wanted to follow that hunch, that maybe he could evade the symptoms of being pressed for time. “I just wanted to not worry that I’m sitting around in the studio jamming until 5am, not necessarily knowing what I’m doing,” said Vile, feeling like he’s frustrated people in that way before. “I knew I’d grown as a musician so maybe I could tap into it easier. THAT WAS MY THEORY. I JUST WANTED TO STAY COMPLETELY BY MYSELF, UNGUARDED, FOR AS LONG AS POSSIBLE.”


Without someone like Agnello pulling the pieces together, Vile needed someone to step into that role. Luckily, he’d been surrounding himself with talented engineers and producers for years. Rob Laakso had been working as an engineer and producer until — after a couple of cameos throughout the years — he joined The Violators

(Vile’s band) full-time in 2011. In fact, recalled Laakso, Vile was the first person to ever pay him to record. Back in 2001 he recorded some tracks for Vile on his 8-track reel-to-reel that serendipitously ended up soundtracking the video intro for lead single Pretty Pimpin’. “Rob is a really good multiinstrumentalist but he’s also a gear and synth nerd,” described Vile. “He can sit there and f**k with tones for a long time, whereas I’m way too ADD for that.” Laakso was installed as the de facto co-producer/head engineer for the record. “I wasn’t the sole producer, like Phil Spector or something,” he made clear. “We just happen to work well together, he’s a good engineer and he’s in my band,” reasoned Vile. “I had these songs I didn’t necessarily want somebody else to play on right away but he’d be there, so I’d be comfortable. There wasn’t any outside person looking at the clock.” Kyle Spence, the Violator’s drummer, also has a home studio called Ronnie Jones Sound where Spence recorded some of the early sessions. Vile also thought it was time he got in on the act, roping in FOH engineer Tommy Joy between tours to help convert his practise space into a recording studio. “We bought Pro Tools and all these things, but I had to say ‘f**k Pro Tools’ for myself. You just have all this I/O popping up and next thing you know you have all these virtual tracks and I don’t know what’s going on any more. “I said, ‘I want to get a tape machine’. Tommy discouraged it but we ended up getting one. Our first experiment was I’m An Outlaw [the second song on the album] and it turned out awesome.” That was before Vile had set foot inside another studio; he knew he was onto something. FEEL NO SAME

While Vile felt good about the process, Laakso had to make a few accommodating adjustments. “At Rancho [De La Luna] I was definitely doing more engineering than playing,” he said. “Which wasn’t how I envisioned it at first. It just seemed like the best way to do it. It would have been fun to be in the room with them.” On the other hand, he prefers “doing guitar parts while I’m driving the computer.” As for Vile’s wont to wile away the hours into the night, Laakso wishes he “could have pushed a little harder sometimes. I can stay up all night if I have to, but it’s not my choice. There was a lot of that on this record, but I don’t think I ever ‘called it’ because I was too pooped. It was a pretty self-motivated record, Kurt wasn’t lacking for songs or material to work on. He was excited to do it; we all were.” The other back-of-brain mental note was sonic consistency across studio hops. “It definitely was something we were conscious of and concerned with going into the recordings,” said Laakso. “Other albums that had been recorded in various locations with different people involved turned out quite well. Whether or not we were always using the same vocal mic, we had faith that it would work out in the end, in part because of the mixers and mastering engineer, Greg Calbi. I like it when albums sound somewhat varied, so long as there’s still a cohesion to them. Some albums sound too


LAAKSO’S CHRONOLOGY OF STUDIOS:.... Red Room — Vile’s practice space. Ronnie Jones Sound, Athens GA — “Kyle’s studio, our drummer.” Rancho De La Luna, Joshua Tree CA Ronnie Jones Sound, Athens GA — “Again.” Pink Duck, LA Thump, Brooklyn NY

Transmitter Park, Brooklyn NY— “No recording that made it, but some editing and a rough mix that did end up on the album.” The Bunker, Venice Beach CA — “Quick stop for work on Life Like Mine.” MANT, LA — “Schnapf’s studio for mixing and some tracking.” Tarquin Studios, Bridgeport CT — “Peter Katis mixing.” Outer Space — Mixing. Sterling Sound — Mastering.

‘samey’ to me. As much as I might love the songs, I can find them a bit fatiguing. “There were a lot of engineers involved. Usually I wouldn’t meddle, but I would respectfully not be shy about calling out bad ideas. I’d bring certain pieces for continuity. But it wasn’t really something that formed any decisions. We wouldn’t refuse to do something because we didn’t have a particular mic that was used on other songs. “There were a ton of mics used on Kurt. I got an original brass capsule AKG C414 towards the beginning of the sessions that everyone agreed sounded awesome on him, but it ended up selfdestructing. I thought the capsule was toast, but that ended up not being the case.” RANCHO RETREAT

A large portion of the tracking ended up happening at Rancho De La Luna. Vile was scheduled to jam with Malian group Tinariwen, who were recording at Rancho. So it made sense to book some recording time for the weeks after. Every song was recorded in a different way, but always with the intention of at least capturing a live performance of Vile’s guitar and vocal to build on. Kidding Around on the other hand, started with a MIDI map, but it was the exception. “Sometimes it’s isolated, sometimes it’s not,” said Laakso. “He was in the room live with everybody during Wheelhouse and there was a fair amount of bleed into the vocal mic. I’d rather capture him being excited in the room, then in an iso booth in his own sequestered corner. The performance would be better. Sometimes we tried doing acoustic tracks in the rooms with the drums at Rancho, which has a fairly small live room. It was a little too ridiculous. Every drum was louder than the acoustic guitar in the acoustic guitar mic. There’s actually a lot of AT 31

When you’re playing an acoustic guitar, you’re listening with your ears. So I just take that distance and move the mic that far away

bleed in Lost my Head there, which is part of the drum sound.” To help control the bleed between Vile’s acoustic and vocal, Laakso usually used a figure eight polar pattern. “But sometimes the vocal will still end up crazy loud in the mic,” said Laakso. “I try to isolate electrics when I can, but not always. Usually it’s just a dynamic mic straight on it. Nothing too out of the ordinary. “He has a bunch of guitars; there was some banjo on this record which was the first time in a while, the Goldtone Dobro on the cover, an old Fender Fender Jaguar. It’s nice that he mostly stuck to the guitars that stay in tune. I remember trying to punch in the bass and keyboard on Wild Imagination. That was a bit of a challenge; it wasn’t dead-on A440 concert pitch. It started off that way but a couple of takes later, not so much. “Pretty much all his acoustics go through an amp. Whether or not it makes the mix is another question. It’s done live, not reamped with a DI. Usually it would go through a bunch of pedals and AT 32

not sound like a natural acoustic at all — a lot of vibrato and delay.” DRIVEN TO DESPAIR

After five studio stops, Vile and co. had a bag full of hard drives and they could feel the pressure of trying to assemble an album from their memory. “Together, Rob and I were responsible for all this music and it was turning into a swamp,” said Vile. After a break at home, Vile got inspired and headed up to Brooklyn thinking they were ready to go through everything. “But all of a sudden it was really hard to do,” he said. “It just seemed a crazy job to finalise it ourselves. It was pretty discouraging.” After that, there were a few more excursions to distract them from the main task; recording in Athens again, playing a gig on the West Coast and recording a little more in LA. Vile: “By then everyone’s looking to me for the answer and it was getting pretty misdirected. We needed somebody from the outside to sift through it. “Rob Schnapf literally reached out at the right

point. He’s friends with Chris Lombardi at my label and by chance contacted him while I was in town. He dropped his whole schedule, so there was a real vibe. We thought he’d just mix the record but then I got inspired and wrote Pretty Pimpin’. We had Stella’s solid backbeat from the beginning, the harmony came quick and he and Rob built it up pretty fast. Rob [Schnapf] also brought in this really cool girl, Genevieve for some backing vocals at the end, and it turned into this still raw, but kind of catchy polished pop song. He did Loser [by Beck], he might as well have a slacker anthem for 2015. My turn!” Schnapf actually had a miscue with Vile early on, so the serendipitous timing of his cold call wasn’t lost on him: “I was working on this Ducktails record a long time ago, and Chris asked me, ‘Do you like Kurt Vile?’ ‘Yeah.’ ‘Well he just finished a record.’ ‘Cool. Thanks for that…’ “It just popped into my head one day, so I texted

MANT SOUND ROB SCHNAPF WALKS US THROUGH HIS DUAL TONE, DUAL CONSOLE.. CUSTOM SETUP AT MANT SOUND. Schnapf: “My main console is a heavily-modified MCI 248B. It used to be a quad mix bus, and I had the idea of breaking it up and making two stereo buses. One is more hi-fi like Sunset Sounds — more forward mid-range — with Jensen 990 op-amps. The other side is vintage — thicker, darker; Neve BA283, Marinair transformers. I have a switch to make them parallel, but if you take the hi-fi bus and drive it into the Neve to saturate the transformers, you get the best of both worlds.”

Chris asking, ‘Hey what’s Kurt up to?’ He texted me back, ‘that’s really weird, he’s in town right now and looking for somebody to work with.’” PIMPIN’ PICKIN’

Vile’s dad, a “bluegrass freak”, bought Vile a banjo as his first instrument. Naturally he acclimated to a picking style, but even when he got his first guitar, Jon Fahey inspired him to continue picking and it’s become a defining part of Vile’s repertoire. Pretty Pimpin’ is the most ‘constructed’ production on the album, and that started with Rob getting Vile to try out his “great collection of guitars,” said Vile. “He suggested I play a ’50s Telecaster, which I don’t usually play, and the acoustic was a Gibson 1954 or ’57 J50. That thing just played itself. I have old Martins which have a bright sound, but this was warmer. “I remember overdubbing the guitar and he’s like, ‘Wow that’s so awesome. Do it again!’ He’d quickly put one on each side, not unlike stuff John would do. He had this vision for it. Pretty Pimpin’

Clockwise from above: Synths for goin’ down (the song) at Thump, Brooklyn. Kurt playing Laakso’s 12-string at Thump. Recording backing vocals on Kidding Around at Rancho. Greg Calbi mastering the album with Peter Katis and Kurt, Sterling Sound NYC.

The modding didn’t stop there though. The EQs are modded, the preamps changed, “the only thing that’s the same is the transformers. “The other console is a 1969 Electrodyne, which I got from Frank Sinatra Junior. The first three ZZ Top records were mixed on one, Beach Boys, and the first Neil Young record. It’s antiquated technology, but the EQ sounds really cool. It’s a really broad two-band EQ that doesn’t sound like you’re EQ’ing but all of a sudden it has more bottom. “They’re jammed together; the Electrodyne has 16 channels, the MCI is 26 with 26 monitor lines. It’s potentially 68 channels, but I’m using in the 30s or 40s.” Schnapf originally only had the Electrodyne. But when he seriously wanted to set up MANT, he knew it wasn’t enough. He wanted to use the computer more like a tape machine, rather than working in the box. “I needed a console and the question was, ‘Do I spend a bunch of money and have to fix something up, or spend not as much money and mod the f**k out of something?’ I went for option B. Let me tell you, I will never do that again! Just buy something that works!” Schnapf is pretty handy himself; repairing amps, modding pedals. He’s in the middle of gutting an Apex 460 microphone, and re-populating it with new internals to see what it can sound like with a U47-style capsule in it. But when it came to modding a console, he deferred to John Musgrave, who mods Neve V series consoles. “He had a company called Mad Labs and used to be chief tech at Capital. He’s a mad scientist, fearless and he never says no. Although sometimes I wish he would!” AT 33

He did Loser [by Beck], he might as well have a slacker anthem for 2015. My turn!

Laakso: “Kurt trying my baritone, which I totally don’t remember even though I took the pic, at Kyle’s place in Athens, GA. Outlaw probably, based on the other instruments.”

definitely had the most help from a producer. He’s just a really good listener.” Schnapf: “Because they’d been working on it for a while, I was just being sensitive to what they already had going on; trying to enhance the process and help it move along. A lot of things aren’t necessarily communicated. Rob knows what Kurt likes, they’ve developed their musical language. That’s why it’s sensitive; you don’t want to turn it verbal because then it becomes intellectual. “It started with Stella, Rob and Kurt jamming it out a bunch of times until we got the arc of the song and the right take.” From there some finger-picking layers were added, electric and acoustic, a solo, and some Moog synth. “It’s orchestrated but there’s a precision to the arrangement. It’s groomed, that’s why it all works together. It’s not just a pile.”

stereo just ends up feeling like mono. There’s not the space for you to feel the stereo of it. Stereo’s okay if the track is simple and open, but I still usually do mono. “One’s got more bottom, the other’s brighter and the guitar doesn’t feel like that. Yuck, f**k that, mono! That’s how you think about a guitar anyway. If I want stereo, I’d rather double it. That’s a better feeling because then you get the bounce of the two takes. “Panning also depends on the song. Sometimes you want it to be way on the outside if it’s percussive and the meat of the song. It depends on what else is going on: where you’ve got to stick them in; which register; is it capo’d or not; all first position open chords?”


“To get the vibe when we were cutting it, he was singing and playing,” said Schnapf. “But we circled back around and did vocals late at night when the mood struck, and he re-cut the guitars. “We’d usually be in the control room recording electrics, except for the guitar solo; he was out in front of the amp with his Jaguar. I use Beyer M160s, on the guitar amps. I also use a modd’ed 57 a lot that’s got a different output transformer and actually does sound cooler. It’s a little more SM7-ey. Just a small variety of amps; the Vox AC30 and an Ampeg Gemini. We used the AKG C414 on acoustic, but then the mic blew up. It was running really hot for some reason and it smoked. “There’s the theory of large diaphragm microphones for finger-picking and smaller diaphragm for strumming. But that’s just a theory, not necessarily a law. “I place it around where the neck meets the body, and as far away as your ear is, but out in front. That’s my theory. When you’re playing an acoustic guitar, you’re listening with your ears. So I just take that distance and move the mic that far away.” Similarly, when it comes to mixing, it’s all about layers. Schnapf: “I always use a compression combination; it might be a Crane Song Trakker and

Schnapf breaks his process down to three ways you can think about an arrangement: “the musical way, the stereo spectrum way and there’s the sonic way — and they all interact. It’s like playing 3D chess. “From a music perspective you might ask, ’what register are you playing in?’ If you have something on the right hand of the piano, so it’s up higher, and you put another guitar part in that same range on the other side, they start talking back and forth to one another. “Or sometimes you stick them on top of one and other, it really depends on what the part is. Is the part supposed to be a texture? Is it a hook? Is it dominant? Is it supportive? Sometimes a part is more of a colour; it’s like the base of the soup, but it’s not the main flavour. “If you were to record a piano in stereo, the right hand is on the right, so if you have another part that is complementary to that, say a harmony, you want it to be over the right hand. If it’s countermelody you might want it to be on the left side so they’re dancing off one another. Those can be ways to think about it.” Schnapf never records an acoustic in stereo though. “If you’re working on a dense track, the AT 34


a UA LA3A, or a Distressor and Tube-Tech. It’s not like you hammer either one of them, you just try and get them to do different things. One can be a little faster and grabs the spiky stuff so the other one can stay parked and take off just 3dB to keep things in check. It just sounds natural. “I keep double tracks EQ’d the same. If I want it to sound different, instead of an EQ I just grab a different guitar. You dial in the zone where the music’s happening and roll out the bottom that’s not really musical information so you can get the guitar to occupy a space without sucking it all up.” IMAGINATION PAYS OFF

Vile also laid down a second song, Wild Imagination at Schnapf ’s MANT Sound. Just a ’70s Maestro organ drum machine — “like a giant shoebox with buttons; here’s the samba, here’s the waltz” — Vile’s guitar and vocals, a simple arrangement and backing vocals. That is, it was exactly the kind of track Vile had been worried about; Peeping Tomboy all over again. He says Schnapf, the unplanned producer who called out of the blue, was the difference between a nightmare recurrence and getting the song on the record. “I was pretty paranoid when I laid it down,” he remembered. “Because it was kind of raw. They all sounded lame to me, so I thought we’d just go from the first take and build from there. He’s like, ‘No man, take two is vibey as hell.’ I listened back and it was paranoia basically. “Even though it’s really stripped down, he really listened and found the right parts. Parts you think would be simple; like the bass outro in Wheelhouse, which is my favourite song. Sometimes the less you play, the louder it’s going to sound. One note is always way better than some fancy extra little flourish that rings. I was always really grateful having him around.” Wild Imagination’s soulful simplicity is a fitting final track. Going on the journey was what Vile hoped it would be. In the end, he had all the time he needed, and just the right people came together in just the right places…

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The sonic skulduggery of Limbo’s composer, sound artist and audio director Martin Stig Andersen. Story: John Broomhall

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Game audio people are still talking about Limbo, and may be for some time yet. Playdead’s debut release rocked the games industry, heralded as a poster child for ‘videogames as art’ thanks to its mysterious and compelling monochromatic visuals. The visuals are only one half of the story though. They’re perfectly accompanied by Limbo’s bleak yet enigmatic, multi-award winning soundscape. It’s a strikingly original take on game audio, especially for a 2D side-scrolling platformer; a style of game that’s more synonymous with Super Mario’s 8-bit bleeps and light-hearted theme music. (You can sing it now, can’t you?) Martin Stig Andersen’s background in acousmatic music, sound installations, electroacoustic performance and video art informed his brave holistic audio vision for Limbo. Andersen decreed that there was to be no ‘spooky’ stylised interactive orchestral score, no character emotes and no narrative dialogue to pollute the ‘nothingness’. That is, he put the kibosh on all the usual tropes videogame composers and sound designers rely on like a crutch. The end result is that footsteps are pretty much the only sounds the main character makes. Andersen feels that in third person games — where the player must identify with a game character — players feel disconnected if the character starts to make noises. And for Limbo, which follows the journey of an unnamed boy in search of his younger sister in an eerie Noir-ish, grayscale environment, it only works if players get sucked in. Even Limbo’s music is so ambient and blends so intricately with the sound and graphics that some reviewers claimed there was none. BRIEF BUT POWERFUL

Andersen’s original brief was simple: ‘not like a videogame’. Which is perhaps not what a typical videogame developer wants to hear. But Playdead left him free to explore this notion of nothingness, and he found the iterative experimentation/decision-making process both engaging and inspiring. Andersen explains: “I learned some interesting things. Trying to make Limbo sound like an old film, I put everything into mono but discovered I couldn’t engage with that sound. It just wasn’t immersive enough. I saw Limbo as such a tiny world, so I was trying to reduce all the sounds to something very simple and thin sounding. I distorted sounds, then afterwards I expanded them again to really spatialise them; almost anti-phase. “I ventured into using antique audio devices; wire recorders, spring reverbs and tape recorders. In linear media you can make your mix from moment to moment, whereas in a game the sounds might be mixed differently every time you play. I discovered that using old machines created a homogenised sound. Running all of my sounds through an old tape recorder made them sit very well together in the mix. “When I put my own bespoke-recorded physics sounds in the game, they sounded too real; the


which made the bottom disappear so it sounded narrow and thin. Because the sounds lost a lot of their main identity and clarity, they suddenly became more generic; I could use the same sounds for a metal box or a wooden box. It all contributed to making the world very small and defined.” PLAYING WITH EMOTIONS

Andersen sees the overall narrative structure built into Limbo’s audio as his biggest contribution: “No one really pays attention to that aspect. For me, the overall framework played a very important part. I was trying to achieve the creation of a world structure with the audio going from the quasirealistic, naturalistic sound you hear in the forest to becoming more abstract and almost transcendent as the boy progresses through the world. “You have the most horribly traumatic moments and the sound suddenly turns into something melancholic, contrasting with what you see. It makes sense to me to say, ‘Ok, so this boy travels through all this violence and I have to respond to this in some way’, otherwise people would just get habituated to it. I WANTED TO MAKE IT FEEL LIKE THE BOY GOT HABITUATED TO THE VIOLENCE RATHER THAN THE PLAYER, LEAVING THE PLAYER TO WONDER HOW THEY SHOULD FEEL. SOMETIMES THE MUSIC WOULD ALMOST REPRESENT FORGIVENESS.

“Limbo is really distinct; something unto itself. Despite real world references, the sound helped to make a ‘limit’; a wholeness to the experience as you move from A to B — a development and an ending.” Small wonder then, that Playdead asked Andersen back to design music and audio for their second production Inside, an equally dark and atmospheric development of Playdead’s creative genius. Those who experienced Inside’s audio demonstration at UK conferences The School of Sound and Develop Conference in Brighton were blown away, with one impressed delegate declaring they ‘may as well give up sound design right now’. This time, Andersen’s audio development is characterised by a blurring of sound design and sound implementation. In other words, he is creating game audio using the audio middleware and game tools themselves. “It’s one thing to create great sounds,” said Andersen. “And another to make them come alive in the game. Creating Inside’s character sounds often required an iterative process where we’d first go and make sound recordings, start developing an implementation strategy for them and then, based on our learnings, go back and re-record in a way that would fit the implementation strategy. “The more the sounds are shaped by various game parameters, the more the game comes alive. We expanded that approach by feeding output from the sound back into the game. For example, THE

Andersen decreed that there was to be no ‘spooky’ stylised interactive orchestral score, no character emotes and no narrative dialogue to pollute the ‘nothingness'



“On a global level, a lot went into implementing custom sound transitions between death and respawn in order to maintain immersion through the unloading/reloading process. That attention to the overall experience by embracing death/re-spawn is something I often miss in games. There’s an intangible dynamic between real-world and gameworld time there. Even though my character dies and I go back in game-world time, real-world time still frames my experience, and I easily get annoyed hearing the same line or music cue over again as I die and re-spawn. However, if I quit the game and get back to it after a few days I probably do want to hear those sounds again. Making a distinction between load and re-spawn, and creating unique mix and music transitions for every situation are integral to Inside’s sound design.”

I’d been playing around with a real human skull in the studio in order to create bone-conducted sound. My goal is that, like a siren song, the gloomy, faint echoes of synths will coax the player forward… to whatever end.


Compared with the Limbo development experience Andersen had a much longer and deeper involvement with Inside, working closely with the team over years rather than months. This, combined with the respect Limbo’s plaudits have won him, has opened some significant doors for audio integration. “It’s allowed me to get at the core aesthetically and technically,” said Andersen. “Doing things that are impossible to introduce later in the process, like prototyping game-play where timing and mechanics are hooked on music or clock time, rather than the usual but much more unstable game-time. It’s great for tight integration between music and game-play but a technical challenge… you have to demonstrate it’s worth the effort early on. AT 38

CREATING UNIQUE WORLDS Andersen: “A lot of the things I do are essentially mash-ups or paraphrases — it arises from working with electro-acoustic music for a lot of years. I can take one sound and it doesn’t really matter where it comes from because I’m not using that sound as it is. I might just extract the texture or colour and then use it to transform another sound. It leads to a slightly un-natural but useful quality allowing me to create an audio world that’s generic and yet unique.”

Is there a plug-in for that?: Andersen has acquired a real human skull to create bone conducted sound — processing sounds through the skull using contact mics. The studio features other lo-fi gadgets for that “sombre, chill quality”.

“Early involvement means sound becomes part of the creative toolset in forming the game’s structure, not a bolt-on. For sequences in which game-play and sound played very well together but eventually became too repetitive sonically, I could suggest changes in the game’s structure. That worked in reverse with the team suggesting sound structure changes, enabling us to create coherent musical build-ups that encompass entire sections of the game.” Just like Limbo, Inside exhibits a unique sonic identity — not so common in today’s games — though according to Andersen, it’s more subtle this time: “It’s the graininess of early 12-bit digital audio hardware like samplers and delays. BY MEANS OF CONVOLUTION, WE’RE RUNNING AN ’80S (THEN) STATE-OF-THE ART HARDWARE REVERB DYNAMICALLY INGAME, WHICH REALLY MAKES THE AUDIO ELEMENTS MELD TOGETHER. Aesthetically, I took inspiration from ’80s

horror B-movies, which often feature a synthesiser soundtrack. I didn’t want any synth per se, just a vague association. I’d already been playing around with a real human skull in the studio in order to create bone-conducted sound. I made a workflow of processing synth sounds through the skull using audio transducers and contact microphones, and then restoring them. The result has a sombre, chill quality. As in those film scores, haunting tones often score something horrible taking place. My goal is that, like a siren song, the gloomy, faint echoes of synths will coax the player forward… to whatever end.”

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Scott: “Prototype of the first Clariphonic, at the time it was called Clarity Control. We did this last-minute measure because the silkscreened 'Clarity Control' panel was no good due to a call from a software company saying, 'Clarity is trademarked.’ My response, being a law school dropout, was that a single word that's used in common parlance is not a legit trademark, it has to be a unique word or word combination (like, oh, Clarity Control), or a word used completely out of context, like calling a soap 'Tide', or a computer 'Apple'. He said, ‘Is it going to be like that?' And the thought came to me to be more clever, so I let it go and later that night came up with Clariphonic, which I reckon is way better. We got the bubbles smoothed out and no one at the trade show noticed it was a sticker.”

UBK got its start modding Empirical Labs’ analogue Fatso, now Gregory Scott is digitally modding the Distressor. We talk about his new company Sly-Fi Digital, and which is better, analogue or digital? Story: Mark Davie AT 40

Holidays were done differently in 1982; quite a bit differently under Gregory Scott’s guidance. 10-year old Gregory’s parents had bundled the kids into the car for a guided cassette tour of Nova Scotia. Its running commentary steering the Scott’s on a course between significant Canadian landmarks… mostly lighthouses. Young Gregory had just requisitioned his first tape deck but had no material to try out his new contraption with. Then it dawned on him that they’d been listening to a tape all day; a tape he could record on. Blithely he popped his parent’s cassette out of the car console as they pulled in for the day. Later that night, beneath the doona cover he slipped a piece of tape over the erase tab and began playing it back, stopping every now and again to record. The next day the Scott family holiday was punctuated with Gregory’s mod — a fresh, inappropriate, look at the Nova Scotian landscape — which his parents, without flinching, listened to for the rest of the trip. LIFE IS ONE BIG MOD

Being a world-class modder of audio gear (UBK Fatso), hardware designer (Clariphonic, Electra EQ and Tweaker) and plug-in developer (House of Kush and new company Sly-Fi), I figured Scott’s childhood would have included more torn-down radios than wisecracks. He’s a joker, for sure, just listen to his UBK Happy Funtime Hour podcast for a taste. But taking a soldering iron to his dad’s hi-fi equipment would have been one hi-jinx too far. “He was upgrading his rig every two years,” said Scott. “Amazing headphones, quadraphonic; whatever was ‘the latest’, my dad was spending money we didn’t have on it.” But he was grateful for his father’s habit because young Gregory was indoctrinated with great sound, not merely sound engineering practice. With hindsight, Gregory’s life looks as planned out as a Nova Scotian sightseeing tour, but his take is that he got “sucked into the life” by doing things he typically wouldn’t. It all started off with a Gearslutz user gathering — an ‘I’ll show you mine if you show me yours’ of audio gear. He’s a selfdescribed hermit, so when he told his girlfriend (now wife) he was planning to hitch a ride along, it raised both eyebrows. When he arrived, Scott recalls the room being divided in half like a school dance — people that seemed younger than him on one side, and a group that seemed older on the other. The older guys were laughing, so he joined them. That night he met Gil Griffith, owner of Wave Distribution and distributor of Dave Derr’s Empirical Labs. They hit it off, and Scott left Griffiths with the open-ended promise, ‘whatever you need…’ A while later he was at a tradeshow helping out on the Wave Distribution stand and noticed people vibing with the ELI Distressor but walking away confused by the Fatso. Surveying the crowd, he realised people were instantly gelling with the Distressor’s distortion characteristics and a bit bemused as to why they weren’t getting the same effects from the more polite Fatso. He took the idea of a more aggressive Fatso to Griffiths, who

sent him over to Dave Derr. Scott had never planned to be a builder of gear, so when he rocked up to Derr’s Empirical Labs headquarters for a Fatso modding how-to, he wasn’t expecting to get more than a few bucks a unit. “We took it to Dave,” said Scott. “And he was like, ‘I don’t even want to know what you did, just sell it on your own and make sure your customers don’t bother me.’” With Derr’s ‘blessing’, he and Griffith formed a business and started shipping units. Scott’s next meeting of fate was when he moved home for a year to house sit his parent’s place and met Kevin. Kevin’s the same age as Scott, and they would have met years earlier in high school if they hadn’t lived in different districts. He’s also a DIY gear obsessive. These days, Kevin does the midlevel implementation of Scott’s analogue hardware designs for Kush. While Scott’s completely at home hacking a device’s output over a bread board, some jumper wires and a rough schematic, he prefers to let a more steadied hand fret about the final layout. “I get something basic cobbled together then give it to this genius and he makes it a million time better, then he hands it to another genius and makes it another million times better than that,” explained Scott. “They send me the first prototype and then I start tweaking from there. I usually do about four different prototypes of a piece of gear.” SLY-FI: A NEW ADVENTURE

Since then Scott has flipped between a mix of hardware and software products, occasionally leafing through his journal of 37 product ideas to find a fresh challenge. His latest are three plugins from an entirely new company called Sly-Fi Digital: the Deflector compressor, Axis EQ, and a one-trick pony mystery box named Kaya. Scott has no real answer as to why he started a new company to launch the new plug-ins, he just chocks it up to instinct; instincts that have served him well so far. And then there’s his bizarre logic, “It makes for a fun job and having two companies is half the work.” … right. DEFLECTOR: ORIGINAL DISTRESSED

The first of the three plug-ins has brought Scott right back to Derr’s front door. The Deflector is Scott’s take on the Empirical Labs Distressor. “I had just come off the UBK Fatso and became really obsessed with what else I could mod. Nothing in my rack was sacred. I loved all the gear I had but no piece of gear is flawless to anybody’s ears. You buy something and there’s gonna be 10 things you love about it and one thing you don’t. “Primarily I wanted to see if I could make the Distressor tonally neutral. It has a specific forward sound that most people love. I wanted to just hear the compression and none of the distorted

I don’t even know if Dave is aware this plug-in exists, and I don’t know if he would care. It’s very purposefully not a Distressor

forwardness it had.” AudioTechnology: “What does Dave think about you releasing the plug-in? At least with the Fatso you had to buy the original units to mod them.” Gregory Scott: “I don’t even know if Dave is aware this plug-in exists, and I don’t know if he would care. It’s very purposefully not a Distressor. It’s definitely in the same family but the Distressor is hardware and this is a plug-in. I know there’s another company coming out with their Distressor soon, and I’m curious to hear it. “My main complaint with all plug-ins, including my own, is that they don’t have the same transient behaviour as analogue does. It’s not that analogue is better, it’s just that the analogue version does a specific thing.” AT: “What’s the main difference?” GS: “It’s all in the transients — half a millisecond to a millisecond of sound at the most. It’s mostly the speed and therefore the frequency of the sound’s leading edge. The waveform is either coming at you or pulling away from you at the speaker, right? So a transient is everything that’s moving forward. Analogue, no matter how transparent or clear it is will slow that. It just shaves it off, ‘you’re not going any faster than this’. It’s got physical limits. No matter what you do in digital, it’s way too fast. To my ears, there’s a form of brightness to the sound that cannot be drawn out. AT 41

(above) Scott: “The first pre-production Clariphonic. It looks official… except that entire front panel is a vinyl sticker we had made the day before the trade show, as evidenced by the air bubbles in the corner." (right) Scott: “If you look closely at the back panel, you'll see the Clarity Control is loaded into a Distressor chassis, ELI was kind enough to sell me a scratch-n-dent.” (far right) Scott: “The Main Gain is born in my head. I had a knob, and a random piece of metal, and I said 'what the hell are you?' After about five minutes of staring at it, I understood.”

“The opposite is true. There’s something about the slowness, and it just gets slower the more coloured gear you’ve got in the path. Analogue sound is slowing down the high frequencies, making the bass come out phase shifted a little ahead in time. Digital can’t do that yet, though it may someday. I actually don’t think it will. I think they’re just mediums that are always destined to sound a little different.” AXIS: DARKER CURVES

It’s this balance between analogue and digital advantages that endears plug-ins to Scott’s desire for modded gear. Case in point, the Axis EQ, which is an homage to another famous piece of gear, the API EQ, but modified without having to swap out physical components. “I modelled the original API and tightened it up digitally,” said Scott. “It’s a lot of components to change without being sure I’m going to even like the result. So it’s really cool to take the modding mentality into digital and hybridise it. “Some things are modded in the analogue gear itself. I’ll then model that and mod it more in the digital realm, either to do things I couldn’t do in analogue or are just too much of a pain in the butt. I don’t want to change 56 capacitors to tighten up the curves, let me just turn a knob to tighten them up digitally.” Other parts require a bit more of a learning curve to get close to its analogue counterpart. Specifically, getting curves to hold their shape at the high end, explains Scott: “Oversampling and a few other techniques will effect what happens at the Nyquist frequency on anything over 10kHz. If you don’t do anything, it rolls off really fast and tilts what is normally a symmetrical curve in analogue. You lose more air the higher you make the filter. There are things you can do that bring that digital curve back to a more symmetrical profile, which results in a smoother tone.” Scott and his unnamed Danish programmer have learnt a lot since building the Electra DSP AT 42

EQ plug-in that came before Axis. It required a lot of DSP to achieve the same curves in the high end as the hardware Electra. Working out how to make the process more efficient without losing any detail required plenty of coding nous, resulting in the CPU demand being halved between the alpha and final release. Naturally, Axis benefitted from the continuation of that pursuit and carries similar principles while being even leaner. Of course, it’s not Electra either, and Scott points out the two main elements that distinguish Axis: “The core algorithm that calculates what frequency I am moving and by how much, as far as I know, that’s kind of the same in every EQ that’s ever been made. There’s just shelves, pads, tricks and tweezes you do to that basic algorithm so that by the time it fully executes the end product, it’s got a different sound. Axis sits between Electra and an API EQ in terms of the tightness of the curve. Electra is super-tight and the API is super-broad. Axis leans towards the API but is in between the two. “I also got aggressive with making Axis dark. A lot of people have been shouting from the rooftops for years that if you low-pass everything at 18kHz it will sound better. So that’s what I did. Before it does any processing and especially before it generates any harmonics, it gets rid of that hash. In digital, if you just do simple things in the right order it can really help you out a lot. “I’m trying to build tools that factor in all these tiny variables. I’m not inventing anything, I’m just arranging parts in a new or different way that I think brings something different to the table.” KAYA: MAXIMUM SAG

The third Sly-Fi Digital plug-in is as close as you’ll get to plug-in pot luck. Send Kaya a signal and there’s no guarantees you’ll come out with something usable. Scott took engineer S. Husky Höskulds’ one-off ‘Pa’ contraption, which is basically an Ampex tube tape deck, where the preamp is wired to the tape head. “It’s absurd,” said Scott. “It’s like the tube preamp is spitting out, I

THE SAMPLING RATE SAGA Scott: “My opinion is that I like them. It’s remarkably difficult to do an A/B test comparing two different sample rate recordings of exactly the same thing. All I know is I used to use 88.2k and now I use 96k with the Antelope Orion. I do a lot of trips in and out of the converters: I record a sound, it goes to the converters, during mixdown it goes out and back in on an insert, and then out and back in again on the mix down. Three passes of conversion through the same converters at higher sample rates doesn’t offend me at all. I don’t ever go, ‘Awww, I’m losing something by doing this.’ At 44.1k, by the third trip it definitely feels like I’ve lost something. “I also think plug-ins sound better at the higher sample rates. If you’re using plug-ins that have oversampling in a lower sample rate session, ostensibly the maths is the same, but it’s still being brought back down to the session sample rate before being passed along to the next thing. So having a high sampler rate all through the session seems to help everything. Both the trips in and out of conversion and the plug-ins. You pay the penalty with the processor though.”

don’t know, 200 times the maximum voltage that tape head ever wants to see.” He whacked a few more controls on the unit to hopefully present some of Pa’s charm to users in a usable way, but it still doesn’t come close to the randomness of the original. “There are things the hardware can do that I couldn’t even come close to modelling. It has a specific sag when you really hit it with a kick drum that we were hoping to capture, but nope, just not there. It’s one of those things that only tubes, this weird thing in the middle with a tape head, and more tubes and transformers can do. It’s not for everyone.” SATURATION OBSESSION

Toying with compression and saturation has been a lifelong pursuit for Scott. It started with juicing up the Fatso’s saturation, and everything since has had a colour to it. Even Axis, an EQ, has a knob for dialling in saturation. He likes to tell a story about how Michael Brauer introduced him to the concept of using compression for movement, not just dynamic control. And it turns out he had a similar formative experience with saturation. Scott: “I used to be heavily into prog-rock when I was a teenager, particularly older Yes from the late ’60s and early ’70s which was heavily tinged with psychedelic and jazz. “I’m not really into it anymore, but for whatever reason I saw a 130g extra thick vinyl re-issue of Yes’s album Fragile on eBay, and thought, ‘What the hell, I’ll grab this record.’ I hadn’t heard it since I was 20 years old, and I could not believe the amount of distortion on it. It was so dark and punchy; I hadn’t heard that sound in so long. I thought, ‘Man that’s beautiful, I need to incorporate some of that into what I’m doing.’ “So that was my a-ha moment. Distortions matter tremendously.” LATERAL LABELLING

Scott’s mind isn’t typical for a technical person, and his plug-ins reflect that. They don’t always hold to

engineering convention. At times, face plates read like a Batman comic, such as the Splat and Crush compression styles on the UBK-1. Other times, like on Pusher, he merely references knobs and switches with pictograms — more magnet please! On Electra DSP, there’s no frequency markings other than a broad operating range for each band. Scott: “The language on the products reflects how I actually think and communicate. It’s not a gimmick to put fluffy words on there because it’s ‘so cool!’ I never think, ‘there’s too much 300Hz on that vocal’ because I wasn’t trained. If I was trained, then my EQs would probably say 300Hz. I have no judgments about it. I just know it’s not helpful for me to have numbers on there. If the snare drum sounds too thuddy, I know where on my EQ I need to tune to cut that out. “I have been strongly considering giving Electra an option where you can click a little switch and it tells you the frequency you’re around. Because I respect that at the end of the day the products are not just for me anymore, they’re for everybody. On one hand I want people to just listen. But if it causes somebody not to use the Electra, I don’t give a crap about the money loss, I give a crap about them missing out on some artistic possibility because I’ve chosen to do something that works for me. That’s kind of a bummer for the world of art. “I’m going to make more concessions in the future towards other people’s workflows, while still honouring the core stuff that matters to me. “Pusher’s interface was an experiment for me. That was the first time I woke up to the notion of making decisions that prevent some people from accessing something the tool has to offer. Who am I serving? “I might offer an alternative skin for Pusher and see what happens if I have a product with two completely different ways to come at it. Give people both options and see which one they gravitate towards and why.” With his knack for out-of-the-box descriptive labelling, it seems like one-knob or application-

specific plug-ins would be right up his alley. What’s in the pipeline for UBK? Scott: “I’ve never used any of the one-knob ones, but I’m familiar with Jack Joseph Puig’s Waves plug-in for bass. I was at a friend’s place and he was using that on his bass and it was amazing. If I was working on electronic music I would probably get the JJP bass plug-in because that man knows how to put sound together and I think it rocks. “I don’t generally talk about stuff I’m working on, but what the hell. I am working on my own take on that concept for artists. These plug-ins use descriptive words, so I’d like to do one for voice, because that’s a tricky one for people; and bass and drums. They’d be the ones I’d focus on because I know those best.” AT 43


Project 46’s Thomas Shaw dishes the truth about delivering a Progressive House track lightning quick in FL Studio 12. Tutorial: Thomas Shaw

Even though it’s one of the most widely used DAWs on the planet, some people still have a weird aversion to FL Studio. I get it, a little. I mean, it’s called Fruity Loops and it’s also got that reputation that everyone cracks it and makes their first beats on it. Nevertheless, when you dive in, it’s really good. I’ve used every DAW and I’m just fastest on this one, which is why I’ve persevered. I use Logic to quantise some live instruments, but I bring all the audio back into FL Studio. It’s also really efficient. The MIDI mapping is simple and with ASIO driver support FL has really low latency — I’m running my whole project with only 5ms of latency. Even though the developer, Image Line, is close to releasing a proper version for Mac, it’s originally designed to work on PC. I see this as a big plus, because you can build a faster machine for less money. Where FL Studio really excels for me is in the sequencing — it’s best of breed. I keep banging on about speed because it can be really important when you’re working full-time on music. For instance, the song we’re looking at here, The Truth, took us four hours and 50 minutes to put AT 44

together. They’re not all like that: Memories took us three years to make, and Last Chance took a year and a half. Sometimes it works, and sometimes it doesn’t. This song worked. Which I needed to finish the song today, because the vocal was being cut again tomorrow, and I was getting on a plane the day after, which also happened to be the album deadline. FRUITY LOOPS GROWS UP

Version 12 of FL Studio was a huge upgrade because of the new mixer. Prior to this release, you couldn’t do simple tasks like select multiple channels, or route all of those channels to a bus. It wasn’t necessarily because Image Line couldn’t program it, it’s just that FL routes things in a different manner. In Ableton or Logic, if you have a track, it’s a dedicated track. You have one instrument in that track, and that’s it. If you want to load a new instrument, you go to the next track. But FL is all sequencer based; the whole thing works like a drum machine. You can put 10 different instruments in the same track if you want. As a result, it’s a little messier because your automation is for individual instruments and sounds, as opposed to actual

tracks. In FL Studio 12, they’ve really tuned up the mixer; selecting is vastly improved and you can easily gang faders. The bussing is much better, grouping is a breeze, and you can send tracks anywhere with virtual patch cables or set it to sidechain via the dropdown menu. It also looks a lot nicer now. The colours are less ‘robust’ and you can scale your GUI to fit any screen, including hi-res ones. Unfortunately, not all the plug-ins I use scale as well, so even though my Razer Blade laptop has a 3K screen, I still run it in HD.

THE TRUTH My background is in singing, which I started when I was in primary school, so I have a soft spot for the melodic. Regardless of Project 46’s style, we’re always going to lean towards songs rather than bangers. This song, The Truth is from our album Beautiful. We worked with a lot of good writers and singers, and I got the piano and vocal from Jovany then added all the extra elements like cymbals, percussion and synth. There are a few key elements to Progressive House. For instance, you’re usually going to have a drop at some point but the first part I try to work out — when I’m listening to the song or thinking about it in the shower — is the main synth lead, and I’ll build the song around that.

THE LEAD The key to Progressive leads is layering. I’d say you need the equivalent of 30-40 voices to get the full widening. I stacked a number of synths to get the sound, including ReFX’s Nexus, Reveal Sound’s Spire and Xfer Records’ Serum. You can make the same sound with completely different plug-ins, it’s all about filling out the full stereo and frequency spectrum. For the first lead I used Nexus’ ‘Big Saw Chords 2’ patch, which is a shitty lead sound but a little lower, which works when you stack it. Spire ‘Lead 1’ is a basic wide saw lead I probably made, and the Serum saw sound reinforces only the top notes. It’s already a lot fatter with just those three. Otherwise it’s just really thin, and no matter how much you master or crank it, it’s not going to sound as wide or full. There’s a piano lead in there to add some spike, and some strings too.

AT 45

THE CHORDS Underneath the lead, I have a classic house piano sound playing the underlying chords. It sounds cheesy, but if you stack it in the right way, it’s like a Big Room version of Deadmaus eighth-note chords. You have to compress it pretty hard though. I shorten the MIDI notes so they don’t sustain through to the next hit. That way it’s a little choppier and they stand out more. Then I over-compress them to make them sit together. The Kontakt piano used for the stabs is the softer sounding of the two, and I sidechained the sharper one to the kick to get it pulsing.

THE BASS The song has those three main components, the lead, the chords, and the bass. The bass is just a stock Nexus overdriven synth bass with some additive EQ at around 1kHz to push the talky sound — it’s like a soft bitcrusher. Usually when I talk to a sound technician, they’ll say, ‘wow, you’ve just added 12dB of EQ in a small band,’ which sounds like it’s a bad thing. Typically if you boost EQ that much, you’re going to distort, but I’m running -10dB on my output, so I’m not even close to clipping. The reason I boost at that frequency is because if I was to boost the actual support chords there, it sounds really weird. But because it’s the high harmonics for the bass, it’ll punch through and support the mids as opposed to squashing them. There are other tracks I’ve done where the bassline becomes half of the mid chords because of this technique. You have to be careful though, because when you’re cranking things you can easily overdo it.

THE BUILD UP The buildup has 10 instruments, which seems like a lot, but compared to other projects it’s not. We wanted to keep this song simple, because it’s a delicate breakdown. I kept it mostly to a snare roll, with a tom hitting on the one underneath, and a rising Sylenth patch. I like Sylenth’s pitchbent sounds because they’re very controllable. I pitched up the snare, and dropped the volume at the same time so it fizzles out right at the very end. At the buildup’s conclusion is a drum fill, which is comprised of the most realistic sounding drums I could find in the Vengeance sample packs. I’ve made my own samples, but their mostly claps and percussion. I placed a phaser on the shaker percussion, automating the depth to produce a rising resonance, and boosted some EQ to make it sound even wonkier, then threw it into the background. It sounds almost like a wind/whitenoise build. I’ve put a few wind effects in there too. Once, I spent a lot of time making all my own wind effects, then realised they sounded exactly the same as all the other wind effects from sample packs. I like to use reverse crashes, because they have their own sound. I make those myself.

AT 46

To get all these different layers just requires time. I’ve made a lot of songs, and this is an example of combining what I’ve learnt by experimenting over the last five years. AUTOMATING THE DROP In the lead up to the drop, I put a high-pass filter on everything except for effects. On the waveform, you’ll see it will be compressed, then it thins out,

before going super thick again on the drop. If you use simple automation in your mix, you can avoid having to drop the volume in your final master. It makes a huge difference on this track, simply because the lead hits on the first beat. Usually leads aren’t hitting at the same time as the kick, so you can just sidechain it out of the way. This one was a real challenge to get it to sound right.

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THE KICK Usually you’d have more click on the kick. But I left it out of this track because it clashed with the synths. People started asking where the kick was, because they didn’t have good enough sound systems to hear that 60Hz bump. Whoops! With kicks, it’s usually about finding where to EQ out the mids to stop it clashing with other instruments, then just making sure there’s some high end. I like kicks that are tighter on top, and looser at the bottom.

MASTERING I mastered half the album, the other half was by Wired Masters in the UK. I always do a WAV export, because it’s refreshing to hear it again, and the project runs more efficiently. I primarily use Izotope Ozone 5. I downloaded v6, but I still use five, because it’s my favourite. Mastering makes a big difference, there’s a lot of general brightening. I re-crisp everything up and use multi-band compression to squash the mids really hard on the lead.

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A really good mastering technique is to use filters, like a low- or high-pass, just to hear what’s going on in certain areas of your mix. Another trick is mixing the vocal in after the compression on the master bus. I find it’s always a lot harder to get the right mix any other way, because once you throw the loudness on, it squishes the vocal. The compression on the master is actually pretty minimal; an average of 1-2dB on any one compressor. It goes from Ozone’s multiband into

Cytomic’s The Glue and Waves’ API 2500. All up, it’s maybe 6dB, which is also what I usually run for headroom. After compression, I just make up the gain on the API — this one was about +10dB. There’s still quite a bit of dynamic range at the end of it all. Love it or hate it, labels like it squashed, and tend to sign off on it if you make it loud.

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Cam Trewin Interview: Neil Gray Photo: Michelle Grace Hunder (Seth Sentry Strange New Past Tour 2015)

Who are you currently touring with/mixing? RÜFÜS, Seth Sentry, The Delta Riggs and The Preatures. What are some other bands you have worked with? Alpine, SBTRKT, Bluejuice, Kingswood, Saskwatch, The Rubens and Stonefield. How long have you been doing live sound and how did you get started? I started out mixing live sound for friends’ bands over 15 years ago. What is your favourite console and why? I have always been an analogue enthusiast, but for the past few years I’ve toured with digital consoles, which has been far more practical. I usually take out Digicos (SD9, SD10 and SD11 on smaller gigs). I find Digico makes the most intuitive digital consoles to work on. Along with Digigrid and UB MADI, I can use all my Waves plug-ins as well as record every show for a virtual sound check the next day. The best thing about those boards is the ability to lay them out with a workflow that suits me best for each show. Favourite microphone or any other piece of kit? The Shure SM57 has to be the all-round work horse and always has been, both in the studio and live. I also can’t work without the Shure B91a; that mic always paints the most accurate picture of inside a kick drum. It’s always dependent on the source, both vocally and instrumentally. I find myself cycling through all brands and types until I find the mic with the best sound. Most memorable gig or career highlight? The show that sticks out the most was a gig years ago at the Espy Hotel in St Kilda for Mix Master Mike from the Beastie Boys. I don’t think I’ve seen a crowd jump up and down quite like that, to the point where I was holding on to the console for dear life and thinking the floor would give away at any minute. I also recently finished a tour with The Delta Riggs who were supporting The Foo Fighters on their recent Australian Tour. That was an amazing experience, just due to the sheer size of that PA. What is your mixing setup now, compared to what it was in 2000? I cut my teeth on analogue boards in the early years, especially the Midas. You can’t beat the immediacy of reaching straight for a channel strip and dialling in that EQ. Digital boards are much more visual and make you fall into the trap of looking at your movements as opposed to just listening. However, the ability to recall the previous nights’ show is invaluable. Additionally, having processing on each channel expands the way I work; it would be hard to go back. AT 50

What are some mixing techniques you regularly employ that you’ve learnt in the last 15 years? My mixing technique is based around the source before anything else; tune the instruments the way you want them to interact in the room. It’s also about feel, balance and not relying on the EQ curve you assume will work. I’m always bypassing the EQ on the channel to hear what my processing is doing. Lastly, take a walk. The best sound is not about the selfish FOH mix position, it’s about making sure the coverage is feeling right throughout the room. The infill is also really important to me, that’s where the hardcore fans are so they shouldn’t miss out on a great mix. In the last 15 years, what are the pieces of gear or features that have come out and been game changers for you? I can’t live without my Waves C6. It’s a modern-day classic tool. I work with a variety of singers, all with different EQ biases and technique. The C6 can be quite aggressive in eliminating those problematic frequencies. The d&b V and J series line arrays, and J and Infra subs, also translate so well. Working with acts that are quite sub heavy, they produce the right depth and low mid clarity I want in the mix. Lastly, MADI devices for virtual sound check and show records have proved invaluable for me in recent years. How have your working methods changed over the last 15 years? Listening more intently and training my ears to recognise frequencies. That helps me work much more quickly now and has really helped both with live shows and in the studio over the years. Any tips/words of wisdom for someone starting out? Be a good listener. Listen to the artist and take on board their direction and vision. When I started out, I would immerse myself in gigs around town and always threw my hand up to help out. I spent years getting paid nothing but still committed and had a positive attitude about it. You also need time on your tools. When I first started in the studio, I would assist on records during the day then would stay well into the night after everyone left, do a Save As and have a crack at my own mix. I’d pump things through the outboard and down the console to get a good feel of what did what. I learnt so much more that way.



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PC Audio Solving USB confusions when connecting your PC to audio/ MIDI devices. Column: Martin Walker

The Universal Serial Bus (USB) has proven really useful for the PC musician, but there are now so many different inter-gear connections that it can also get confusing. USB devices essentially start out as one of two types. Those with rectangular-shaped USB Type-A ports are host devices, including PCs and some Macs. They’re also found on USB hubs, and in many cases are also able to supply USB 'bus' power to devices plugged into them. Meanwhile, the humbler USB peripherals including many audio and MIDI interfaces, USB-equipped keyboards/synths, and boring devices like printers and scanners sport squarish USB Type-B ports with two slightly bevelled top corners. As electronic devices shrunk, Type-B ports were too bulky to fit, so a miniaturised USB Mini-B connector started to appear on many portable audio recorders, audio/MIDI interfaces, digital cameras, phones and so on. The most popular amongst audio manufacturers is the Mini-B 5-pin version. Confusingly there is also a 4-pin version, but thankfully the two are different sizes and shapes so you can't accidentally plug in the wrong cable. A set of even smaller 'micro' sized B-connectors appeared for modern mobile phones, but these don't tend to be found on many audio/MIDI devices. So, the majority of USB scenarios start out requiring a USB cable with a Type-A plug on one end to plug into your computer or hub, and a Type-B plug on the other that plugs into your peripheral device. I GOT THE POWER!

A to B connections neatly avoid possible overloads or electrical damage, since power can only travel from a Type-A to Type-B port. Where you can run into problems is if you attempt to connect two computers together via USB to transfer files between them or create a network to share your internet connection. Yes, you can buy Type-A to Type-A cables, but if you plug both ends into different computers this could either burn out their USB ports or even damage your computer power supply. For this application you instead need a special more expensive 'bridged' or 'USB networking' cable. AT 52

Even with an apropriate A to B connection, it can be a bit hit ’n’ miss powering audio/MIDI devices, since only so much juice will be available from each Type-A port. Mains-powered synths/ controllers/interfaces won't need any power from your PC or USB hub, so they’re fine. If you're connecting your USB-powered audio or MIDI device to a desktop PC, chances are there's plenty of power available too — a typical Type-A USB port will supply up to 500mA. It's when you connect an Audio/MIDI device to a battery-powered laptop/notebook (or tablet) that things can become hit and miss. If there's not enough power available from that host then it simply won't power up or be properly recognised. With all battery-powered host devices, adding a powered USB hub between it and the USB peripheral will invariably let the latter be recognised, if that doesn't defeat the object of your setup being battery-powered. MULTI-HOST SHENANIGANS

Where things can get really confusing is with the new breed of Multi-host standalone USB Audio/ MIDI devices. As well as traditional audio/ MIDI inputs and outputs, they can also serve as a glorified central hub with complex routing options between all your MIDI instruments/controllers, mobile devices, and a simultaneously-connected Mac and PC — this can be a truly liberating experience! It's easy to get thoroughly confused by the possibilities at first, but the important thing to remember is that Type-A ports (often labeled Host) on such standalone MIDI/Audio interfaces are the ones that may supply power to peripheral devices, so should be connected to MIDI instruments and controllers. Any Type-B ports are designed to be connected to PC or Mac computers via a standard USB B to A cable, or via a special (and often supplied) USB B to Lightning cable to connect an iOS device. If you plug in the Multi-host mains PSU, one of these USB Type-B sockets may also be able to supply sufficient juice to charge a connected iOS device, so you no longer need to periodically unplug it when its battery finally runs down. Another Type-B port on the same multi-host

device may instead accept USB bus power from your PC or Mac when you don't want to use the Multi-host PSU. You need to carefully read the manual to find out what options are on offer. Such Multi-host devices also generally offer internal audio/MIDI patchbays controlled by Mac/PC software, so you can route audio and MIDI signals between your various computer and MIDI-based gear. The possibilities can be mind-boggling! USB 3.0 AND C-TYPE

Since I've been discussing the various USB connectors, I ought to conclude by mentioning the other USB standards you may stumble across in your travels. Those new-fangled USB 3.0 and 3.1 SuperSpeed devices with their own modified Type-A and Type-B connectors still follow the same rules as those designed for USB 1.1 and USB 2.0 duties, although we don't see very many audio interfaces still featuring those ports. The most recent USB C-Type ports potentially confuse matters by featuring the same roundedrectangular reversible-plug connectors at each end of their cables, which can plug into either a USB-C host or peripheral with no worries about potential damage. However, USB C-Type devices can also be connected with devices of all the previous USB standards using a cable with USB C-Type at one end and Type-A or Type-B at the other. Standards, eh?

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Interface Roundup




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FOCUSRITE CLARETT 8PRE Audio Interface Another Thunderbolt interface hits the market, but does this Focusrite unit occupy rare ‘Air’? Review: Preshan John

DO-IT-ALLS — Two XLR/instrument combi inputs on the front panel give easy and quick access when hooking up a mic, line input (keyboard) or guitar/ bass direct, with Line/Instrument input switchable in Focusrite Control.

MAKING GAINS — These eight pots control the 57dB of gain for each built-in preamp.


AIRTIME — The row of yellow lights show you which channels have ‘Air’ activated; a circuit which models Focusrite’s ISA preamplification.

PRICE Expect to pay ~$1500 CONTACT Electric Factory: (03) 9480 5988 or

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PROS ‘Air’ feature sounds great Very stable in use Quality feel & construction

CONS Only one Thunderbolt port

MIX IN THE CANS — You can create a separate mix for each of these two headphone outputs using Focusrite Control.

HEAR WITH YOUR EYES — LED metering on the front panel for the eight builtin inputs. Great for gaining up a bunch of mics quickly, without having to look over at your screen.

SUMMARY The Focusrite Clarett 8Pre joins a growing list of super low-latency Thunderbolt interfaces, but the ISA preamp-emulating ‘Air’ feature gives the 8Pre more than just one flavour. Not something you get from a lot of interfaces.

Depending on which way you spin it; I’m either straight-out biased or the perfect person to review Focusrite’s new Clarett interface range. Let me explain, and you decide: Currently, I’m the user of not one, but two Focusrite interfaces — the small and portable Forte, and the 1U Saffire Pro 40. The Forte is a great-sounding portable unit I carry around for small sessions — tracking acoustic instruments, vocals, etc. The larger Saffire Pro 40 has been the hub of my home studio for the last five years. It’s been a pleasure to use and several memorable recordings have come out of it. It’s starting to show its age though, so I’m on the lookout for a worthy upgrade option that won’t change my workflow too much, or break the bank. It’d also be nice to progress from the Firewire 400 protocol. I won’t deny that a Thunderbolt-equipped, similarlysized contender from the same brand seems like a natural fit. On the flipside, I’m also perfectly placed to decide whether it’s a worthy upgrade. The final piece of this Focusrite puzzle is my ISA428 MkII rack of ISA preamps. Interfaces can be a bit same-ish these days, and the distinguishing feature of the Clarett range is its Air. A circuit that’s supposed to mimic the sound of an ISA pre. So yeah, I know how that’s supposed to sound. The big questions for me are: Will the Clarett be a substantial upgrade from the Pro 40, and does it fulfil this lofty, high-on-oxygen goal of emulating Focusrite’s ISA heritage? GAIN SOME CLARETTY

The Clarett line is an entire product range that could suit the needs of a beginner or an established project studio. The family consists of four interfaces: the 2Pre, 4Pre, 8Pre and 8Pre X. All of them record up to 24-bit/192k and are expandable via ADAT, even the compact 2Pre. They’re also Focusrite’s first Thunderbolt-capable interfaces. Just as well, given that brands like Apogee, MOTU, UAD and Zoom have already released Thunderbolt interfaces of their own. As a part of the package, you also get Focusrite Control routing software and the Focusrite Red 2 and Red 3 plug-in suite. We were sent the 8Pre for review; the secondfrom-top model with 18 inputs and 20 outputs. On the surface, you’d be forgiven for thinking the Clarett 8Pre is a red, Thunderbolt’d version of the Saffire Pro 40. A glimpse at the front and back panels and all you’d think Focusrite managed was a change in colour scheme and the addition of one Thunderbolt port (yes… just one). But closer inspection reveals slightly more meter resolution, the addition of a wordclock Out port (no In), a bunch more LEDs, and a few physical buttons relegated to software. The big deal about the Clarett range, apart from Thunderbolt, is the preamplification. The pres in the Pro 40 aren’t amazing — the high I/O count was the star attribute of the 1U unit when it came out.

But with the new ‘Air’ feature, you’re supposedly getting preamps modelled after the transformerbased ISA pres — something that, if achieved well, will make it an attractive proposition.

glad you can turn it off. I found it to be entirely usable addition to the interface. It’s like having two preamps in one — who wouldn’t want that?


If there’s one thing I appreciate about an interface it its flexible and intuitive routing software. I generally leave all headphone mix duties in the hands of such software as it means minimumlatency monitoring for the musos, reserving the DAW for tracking alone. The Clarett interface's routing software is called Focusrite Control and bears little resemblance to its Saffire, Scarlett or Forte counterparts. It isn’t the simplest GUI to get your head around, but it makes perfect sense once you’ve nutted it out. The idea of adding input channels to each individual mix felt strange initially, as I’m accustomed to seeing all the inputs at once. Though once I had adapted to the look I appreciated the decluttered mix tabs. There are a number of useful presets you can load up, and the snapshots feature lets you save your I/O configuration — invaluable for when the band says they want to redo yesterday’s session. The bundled 64-bit AAX, AU and VSTcompatible Red 2 EQ and Red 3 compressor plug-ins sound great, and worth £229 no less. Their transparent character makes them well suited to sources that require unobtrusive, clean treatment.

The classic Focusrite ISA microphone preamps are a staple in plenty of recording studios. First appearing in the ’80s, the ISA was birthed when The Beatles’ producer Sir George Martin commissioned Rupert Neve to create a console for AIR Studios, Martin’s recording space in London. My foray into the world of Focusrite preamplification started out with the ISA One — which I liked so much that I bought the ISA 428 MkII. They’re what some call a ‘vanilla’ preamp; a bit flavourless compared to the ‘chocolate-swirlswith-a-cherry-on-top’ counterparts you might find racked up next to them. But, in part due to its Lundahl transformers, the ISA preamp has a subtle character that just seems to work for most things recorded through it — it’s an excellent utility preamp that stacks very well in busy mixes. So the idea of the Clarett 8Pre having eight ISA-esque pres in a single 1U interface is appealing. But can you get that transformer mojo out of a transformerless mic amp? The proof, as they say, is in the (vanilla) pudding. Naturally, I decided to compare the Clarett preamps with ‘Air’ switched on to my ISA ones. Easier said than done. Without wanting to fry either input with phantom power and a Y-cable, or change the load, I resorted to recording several acoustic guitar tracks with a single mic using both pres; layering a stack of Clarett recordings next to a stack of ISA ones. The hope was that multiple layers would magnify the differences between the two without biasing a particular performance. After excessive A/B-ing, the ISA tracks came across slightly more ‘solid’ than the Clarett pres, most noticeably in the low-mids. The ISA’s high end was creamier, and the overall sound was a little more elegant and composed. However the difference was minute, and most importantly, the ISA DNA was certainly evident in the Clarett preamps. Kudos, Focusrite. NEED SOME AIR?

The ‘Air’ effect is individually switchable from Focusrite’s Control software. It’s unclear what the actual circuit change is, I suspect it has something to do with jacking up the impedance, but there is a satisfying ‘click’ of a relay when toggling it on and off. The effect isn’t subtle, but it’s not overstated either. It introduces a boost in the high end that works beautifully on most tracks. Acoustic instruments are injected with a little more sparkle and it provides vocals with a pleasing sheen. There may be the odd occasion where you find ‘Air’ doesn’t match a mic or source so well, and the high end gets a bit gnarly; in which case you’ll be



For what it’s worth, the primary red colour of the interface is very appealing. Sporting a brushed steel finish, it looks pretty striking sitting in a rack. The build quality is excellent too. There's a good weight to the unit and nothing about it looks or feels cheap. Each knob has a smooth, resistant feel to it. The one things that bugs me is the single Thunderbolt port. Unless your computer has more than one of them, you rely on daisy-chaining to hook up two or more devices at a time. Thankfully the newer Apple iMacs and MacBook Pros ship with two Thunderbolt ports, but for those using older computers, you’ll be limited in some ways. If, like me, your external monitor connects via the Thunderbolt port, then purchasing a Thunderbolt hub becomes your only option — and even the most affordable of these will still deepen the dent in your bank balance already caused by buying the interface. That niggle aside, Thunderbolt is lightning fast, and it’s no different with the Clarett. Focusrite quotes some pretty compelling roundtrip latency figures you can find on its site. Even going so far as to put an N/A marking on its 32 sample buffer test at 96kHz in Pro Tools 11. In our own tests, there was some slightly weird software compensation going on, whereby a signal sent via an analogue output would arrive back into Pro Tools via an analogue input six samples AT 57

before it left. Typically there would be some delay here, caused by the A/D and D/A conversion. The same anomaly occurred on two different computer systems. Focusrite is obviously trying to compensate for any latency where it can. Latency was super low overall. We measured a regular addition of 58 samples to the buffer per roundtrip, using Pro Tools 12 on a four-year old Macbook Pro — 0.6 of a millisecond at 96k. In other words, totally negligible. MAKING TRACKS

I took the 8Pre on the road to track some demos with a five-piece band consisting of a lead vocalist, two electric guitarists, bassist, drummer and keyboardist. We decided to record live; so 14 inputs all up, with an extra eight preamps coming from my UAD 4-710D/ ISA 428 MkII combo into the Clarett via ADAT. I’ve experienced the occasional glitchy connection using the Optical input with the Pro 40, but after setting the clock source to ADAT in Focusrite Control, the Clarett-centred setup was rock solid for the entire recording session. And the lightning-fast Thunderbolt connection wasn’t phased by the high track count or my ageing laptop. The tracking process was smooth sailing. Setting levels was a breeze thanks to the LED meters on the front panel. The mic inputs were immaculately clean and noise-free, even when turned up to nearly maximum for the vocalist’s Shure SM7b. Creating AT 58

two independent headphone mixes for the vocalist and musicians was easy using Focusrite Control. I wish the GUI wasn’t so monochrome though… Switching between Control and Pro Tools felt like the computer’s saturation setting was broken.

HEAR AIR IN ACTION After tracking some demos for the band, the drummer played a few grooves for me while I switched in the ‘Air’ settings for the kick, snare and overhead mics. Check out the untouched audio samples at to hear ‘Air’ in action for yourself. The change in sonic character is very noticeable in this four-mic recording, especially in the hats and snare.

While packing up after the session, I was musing at how well it all went. A lot of things could’ve gone wrong. If it’s not risky enough taking a brand new interface to a recording session, I had also just upgraded to Pro Tools 12 without much chance to test it with the Clarett. But even with the hasty prep time, everything just worked, and it felt good. CLEAR CONTENDER

So to answer my original question: yes, the Clarett 8Pre makes for an outstanding upgrade option

from my Saffire Pro 40; and here’s why. Firstly, the ‘Air’ feature deserves the hype — it’s very musical, and the option to switch it in and out effectively doubles the character of the interface. I would consider buying it for ‘Air’ alone. Secondly, it’s solid. Yes, it’s built well, but more importantly the Clarett performed with absolute stability every time I recorded with it. Granted, it was only tested for a few weeks; but the fact that it didn’t miss a beat from the moment I plugged it in to the moment I’m writing this paragraph says something. And finally, it’s got Thunderbolt. The FW400 connection on my Pro 40 is so yesterday — I even need an adapter to connect it to my laptop. Thunderbolt makes Firewire appeared pedestrian when it comes to latency. But if only it had one more port! What if you’ve never owned an audio interface before? The price tag may be a deterrent, but acquiring any of the Clarett models will set you up for an enjoyable recording experience; not to mention your tracks will sound great. If you’re considering an upgrade from a smaller desktop interface, the 8Pre or 8Pre X would be fantastic candidates to get some more I/O. And for those looking for less I/O, but worry about losing features, Focusrite hasn’t skimped — you even get ‘Air’ on its tiny 2Pre.


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PRESONUS STUDIO 192 USB3 Audio Interface Lots of I/O, quality preamps, monitor section, DSP… PreSonus is on familiar ‘great value’ turf. Review: Brad Watts

Four backlit buttons address 48V power to each preamp input, along with talkback, mono monitoring, and output dimming/muting.

The internal talkback mic can be replaced by an outboard mic connected via any of the mic preamps.

PreSonus’ remote controllable XMax solid-state preamps (60dB gain, 118dB dynamic range) with an added separate digital volume-control circuit for digital recall without affecting the sound quality.


I’ve been a fan of Presonus’s audio offerings. I keep a set of the company’s preamps on hand (the now-discontinued Digimax FS), simply because I like the sound of the ‘XMAX’ pres and the versatility of the unit. At the time it was rare to find Class-A, discrete, high-voltage preamps and 96k converters in such a form factor, and at such an agreeable price. It’s true, PreSonus leans toward the budget end of the audio world with equipment punching well above its weight in terms of audio reproduction and capture, but this is exactly the PreSonus ethos: top-shelf results made affordable. This ‘quality for all’ philosophy continues with PreSonus’s latest recording kit, the Studio

PRICE Expect to pay $1399 CONTACT Link Audio: (03) 8373 4817 or

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Input levels are represented via eightsegment LED meters, as are the main left and right output levels, and blue LEDs atop the mic level input meters signify 48V power activation.

192 Recording System. As one would assume, the Studio 192 will cater to sample rates of 192k via the analogue inputs, and the unit can be expanded using S/MUX-enabled ADAT optical connections, which will of course reduce your capture sample rate to 96k – which ain’t half bad. That’s an additional 16 channels at up to 48k, or an additional eight channels at up to 96k. Ins and outs, all up, amount to a healthy 26 inputs and 32 outs, and the unit achieves all this via USB3.0. Speaking of which, the PreSonus support notes mention there to be issues streaming audio using a USB3.0 cable when using OSX 10.9, 10.10, and 10.11. The reported ‘fix’ is to use a garden variety USB2.0 cable, which apparently results in

PROS Sounds great Complete monitoring system Very expandable Terrific preamps USB connectivity On board DSP Cheap!

CONS No MIDI I/O Buttons ‘out’ when they’re in Level jumping mono switch

Two headphone amplifiers with independent outputs and level controls.

no loss of functionality or performance. Oddly, I had no issue using the supplied USB3.0 cable on OSX 10.10.5. Maybe this is because my Mac is a bespoke, Frankensteinian, hand-crafted monster… maybe it’s not. But it worked for me regardless. Some would question the use of USB3.0 over Thunderbolt, but with PreSonus’s Studio One DAW platform being cross platform, and with Thunderbolt-endowed PC motherboards being thin on the ground, it makes sense to stick with USB as the streaming conduit. Minimum OS requirements are OSX 10.8, or Windows 7x 64/x86 SP1, but cross reference this with your DAW’s OS requirements.

SUMMARY More than a USB3.0 192k-capable audio interface, the ‘centre section’-style features turn the Studio 192 into a highly flexible command centre. Powerful and great value.

16-channel ADAT optical in/out (eight channels at 88.1 or 96k), coaxial, stereo S/PDIF I/O; and BNC wordclock I/O.


Nowadays, it seems standard practice to incorporate monitoring and talkback features into this level of interface. As such, the Studio 192 includes typical control room features, including the ability to switch between three sets of monitors, although this is only possible via the included software control panel. Connection of additional speakers involves using some of the eight balanced TRS output jacks. These can, of course, also be used for monitor mixes, and there are dedicated TRS outputs for main monitors. For actual tactile control, the front panel of the unit offers a large main output level control pot, with a single gain level control flanked by two left/right buttons. These buttons scroll through the eight preamp inputs, with the gain control adjusting the preamp displayed via the front panel LEDs. A ninth ‘c’ setting provides level control of the built-in talkback mic (‘c’ stands for ‘Comms’). Incidentally, and far from uninteresting, the preamp levels and individual 48V power for each mic pre can be set from within your DAW using MIDI controller information – MIDI Channels 1 to 8 correspond with each pre, with CC#7 assigned to gain, and CC#14 for phantom power. This is exceptionally useful as you can snapshot your preamp levels between projects. Knowing this, if the Studio 192 can receive MIDI information, surely there could have been some MIDI I/O included. Doesn’t anybody use those connectors anymore!? Breakout cable anyone? HANDS ON THE WHEEL

Four backlit buttons address 48V power to each preamp input, along with talkback, mono monitoring, and output dimming/muting. The dim/mute button provides both operations – holding down this button will mute the main outputs, while momentary prodding will dim the main output by 20dB. I was a tad confused by how these buttons operate. Both the talkback and mono buttons activate when releasing the button, rather than activating at first push – which felt somewhat counterintuitive. Also, the mono reproduction sounded noticeably louder than the stereo signal – a substantial annoyance [PreSonus are now aware of the problem and promise a fix by January – Ed.]. I’ve covered off the rest of the front panel on the

Six mic/line inputs complement the two front-panel mic/instrument inputs. Eight balanced TRS outputs, balanced stereo main outputs.

photo markup but the preamps are worth further comment. I’ve already pointed out my fondness for the Class-A XMAX preamps. The spec has been incrementally improved upon in some respects compared with the Digimax FS pres, including an extra 5dB of gain (up to 60dB). PreSonus obviously believes it had the design right many years ago and has used the XMAX design over many of its products. Most notable in its insanely popular StudioLive mixers. Around the back, you can consult the photo to see what’s on offer. The power is supplied to the rear via a ‘lump in the lead’-style power supply and the power switch resides on the front panel. The unit won’t remember its power state. For example, if you power down your entire system from a main power switch, you’ll need to remember to individually power-up the Studio 192. A minor annoyance. FAT CHANNEL DSP!

The first 16 inputs of the Studio 192 offer PreSonus’s ‘Fat Channel’ processing. This DSP includes phase reverse, a gate (which includes frequency gating), a compressor (with integrated output limiter), and a full parametric four-band equaliser with an additional high-pass filter. That’s a lot of processing across 16 channels, and should provide ample control for recording most bands and ensembles. The processing can also be recorded directly to your DAW by flicking each channel to post-send. In addition to the channel processing are two master effects. A reverb processor with nine algorithms (encompassing various hall reverb sizes, ambience, and plate reverb), along with a separate processor for delay effects providing mono, stereo, filtered and ping-pong delay settings. The editing screen for this includes a tap-tempo button. It’s basically a first edition StudioLive built in, accessible via software. If your DAW of choice is Presonus’s Studio One, all this DSP is accessible via the DAW itself, rather than flitting back to the UC surface software. In fact, the controls show up right alongside the Studio One mixer controls – über integration! DSP dynamics and EQ directly from within your DAW? Excellent! Presonus provide a free version of Studio One Artist with the Studio 192, so if you’re without a DAW (or sitting on the DAW fence) you can kick off from square one with the Studio 192 package.

UC Surface control panel: software is also available as a free iPad application or as a touch responsive app for Windows 8 touchscreen computers.

I’ve no complaint regarding the sound of the DSP. The compression performs as expected, and can pull off ‘character’-style compression if you replicate settings from mainstay compressors. The EQ is also very good, and with so much control over bandwidth you can set up traditional EQ curves easily, or get completely surgical. The delay effects are perfectly fine – the array of parameters isn’t extensive, but they do the job admirably. The reverb is functional. It certainly isn’t Lexicon or TC electronic territory but it’s certainly quite usable. And with only three reverb parameters to tweak (decay, pre-delay, and early reflection levels) they’re designed for immediacy. Effects settings can be saved and recalled. There are plenty of presets to start with, and to augment, save, and recall, either within the UC Surface software, or from within the Studio One DAW if that’s your chosen platform. However, what’s possibly most important regarding the onboard DSP is it can be applied live, with none of the latency one would experience when adding such processing via a DAW. AT 61

DRIVE BY WIRE Supplied alongside the Studio 192 is a software control panel and mixer. Dubbed ‘UC Surface’, the software is also available as a free iPad application or as a touch responsive app for Windows 8 touchscreen computers. Aesthetically the software follows the current penchant for ‘you can have any colour so long as it’s grey’. Like most audio interface mixer control panels, UC Surface is primarily designed for configuring multiple mixes via the Studio 192’s additional outputs. Up to eight separate stereo mixes can be configured, and these can emanate from the onboard eight analogue outputs and the first eight ADAT outputs – likely more monitor mixes than you’ll need but the potential’s there if needs arise. Outboard sends? Backup recorder? There’s enough outs to do it. These features I’d consider foundational to any native DSP-based interface, and historically have usually been supplied to provide direct monitoring from the interface, thus avoiding the round trip time-lag associated with native-based DAW applications. Nowadays, modern CPUs, DAWs, and interfaces can easily reduce this latency to less than 5ms from input to output — so the problem is less of an issue. What’s extra cool about the UC Surface software is its access to the Studio 192’s DSP, which I cover off elsewhere this review.


It’s difficult to fault most modern audio interfaces’ audio fidelity. We live in an era where pretty much everything you buy sounds great. The Studio 192 certainly won’t disappoint on this front, and neither should it as a flagship interface. Presonus has gone the full monty with spec’ing up the 192, utilising a Burr-Brown chipset for A/D and D/A conversion. Not so long ago you’d only find this componentry in audio converters costing at least twice as much as the Studio 192. In fact, the dynamic range of both the A/D and D/A processing is up at 118dB (A-weighted) — that’s up there with interfaces costing four and five times the price of the Studio 192. THD+N figures don’t stack up nearly as well, however. At <0.005 across all inputs and outputs, this figure doesn’t come close to the <0.000x figures touted by the big boys in analogue conversion, and by that I’m referring to the flagship Apogee devices, those from RME, and the Prism Sounds of this world. You’ll move your distortion figures that extra decimal place when you spend many AT 62

more thousands of dollars. As things stand, the Studio 192 competes favourably with units such as the Apogee Ensemble and the UAD Apollo units equipped with similar I/O, all the while landing in your rack with far less insult to your credit card. So would I own one? Yes indeedy Sir, I would. Despite a few micro niggles I’d be jumping on a Studio 192. Not only does the unit sound very good, it’s also expandable to a rather full 24 inputs, and will function on a multitude of platforms and computers due to its relatively standard USB connectivity. The Studio 192 has an enormous amount on offer at a very attractive price.

Preamp levels and individual 48V power for each mic pre can be set from within your DAW using MIDI controller information …exceptionally useful as you can snapshot your preamp levels between projects


for flawless audio performance

OPTIMISED DYNAMIC RANGE for perfect recording levels


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DSP-Equipped Audio Interface Review: Mark Davie

Price: Expect to pay $5599

This is now the third time we’ve reviewed a UAD Apollo unit, and the second time for me. My last outing was with the Apollo Twin, a handy introduction to the Apollo range, but I’ve been eager to try the system again with a few more inputs. The Apollo 8p is the middle child of the secondgeneration ‘blackface’ Apollos — there’s also the standard four-preamp, eight analogue-in Apollo 8, and the preamp-less, 16-channel Apollo 16 converter. The 8p has eight channels of conversion, and eight Unison preamps to go along with them. If you’re unfamiliar with Unison technology, UAD has basically taken a vanilla IC preamp and used its own modelling technology to juice up the harmonics. More than just an addition down the insert chain, when applied in UAD’s Console 2.0 application, the Unison preamp emulations — Neve 1073, API Vision CHannel Strip, etc — take over the gain staging of the preamp. Rather than just adding harmonics, it’s like you’re interacting with the hardware. The onboard preamps are very usable but flicking on any of the emulations instantly adds some extra presence. Each of the emulations added AT 64

harmonics that helped get the sound off the couch and coming forward out of the speakers a bit more. There was never a time I didn’t want to add a Unison flavour, and they always stacked well. The Apollo 8p is designed specifically for Unison die-hards; considering you’d probably go for a classic Apollo 8 if you had some external mic preamps, you were intent on using. The 8p has plenty of grunt for processing UAD’s suite of plug-ins with a Quad Core DSP inside. Unlike the Twin, which debuted without the ability to serve as a monitor controller for other Apollo units — it can do it now, but was a feature many were expecting at launch — it seems as though Universal Audio has covered every angle prior to the 8p’s release. Yes, you can combine it with previous generation Apollos — provided you add the Thunderbolt option card. Yes, the line inputs do bypass the mic preamp stage. Yes, there are two Thunderbolt ports on the device so it can be used as a pass-through; though still no Thunderbolt cable included. Come on now! You do sacrifice a hair in the digital audio department. S/PDIF is nixed on the 8p to make way for the extra four combo connectors on the

Contact: CMI Music & Audio: (03) 9315 2244 or

rear, though you still get two banks of ADAT and wordclock with a built-in termination switch. It’s probably mostly a pain for anyone running an external stereo DAC via S/PDIF for their monitoring, but given the quality of the Apollo’s outputs, I’d question that decision anyway. There’s also no MIDI input on the interface; however, that seems to be de rigueur these days. I do really like working with the Apollo. It would be a serious consideration for me if I was on the lookout for a stable, great-sounding interface, that packed a lot of I/O into one rack space. I’ve reviewed a lot of interfaces, and not many get through the weeks of testing without at least some niggles; whether its OS incompatibility, a weird firmware glitch, the software needs a reboot, or something else along those lines. The Apollo 8p, and the Twin before it, have been entirely glitch free, a complete breeze to set up, and just feel like they’ll keep on trucking. What’s more, the Console 2.0 application is refreshingly easy to use. There’s plenty of visual feedback when you roll over buttons, and you can access everything from the Overview screen. While you can drill down into different tabs, you

don’t have to, which is a welcome change from other systems. Plus, the whole thing looks darn sexy. It complements the hardware’s well laid out front panel, but to be honest, I barely used the hardware controls — except for headphone and monitor levels, and switching between alternative monitors (which you can now do via the front panel). Console 2.0 is simply that nice to work

in. There are three new buttons on the hardware I should probably mention: the first switches between input and output metering, the second switches to a designated alternative monitor output (that you can configure in Console 2.0), and the third is a function button you can assign to a third monitor output, Dim, or Mono — your choice. Of course, I could go on about how there’s

Pick the difference: UA’s model (above) meticulously emulates the original.

more dynamic range in these ‘blackface’ versions (3dB) and more output but we already thought the original Apollo was stellar, and using and listening to the 8p there’s no reason to doubt it’s even better. Besides, you’re really here because you’re thinking about getting into the Apollo world and all those UAD plug-ins. It’s not a cheap place to be, but boy is the whole package very appealing.

The envelope follower on UAD’s second H910 delay line responds to the original signal, altering its own pitch based on its envelope. It also comes with a graphic representation of the HK941 keyboard controller you can use to change the pitch.

ARE UAD & EVENTIDE IN HARMONY? Reinforcing their claim of modelling classic gear as accurately as anyone’s ever done it before, UAD’s version of the Eventide H910 Harmonizer comes with its own proof. The first pages of the UAD plug-in’s manual opens with Eventide’s original marketing collateral for the H910 from 1976. The exact model is right there in the picture. While playing around with UAD’s version, I also downloaded Eventide’s recently released Anthology X bundle. In amongst the newer releases like the UltraChannel, UltraReverb, Octavox and Quadravox are some of its legacy plug-ins re-released for newer formats like AAX. This includes the Clockwork Legacy series which has Eventide’s own emulation of the H910 Harmonizer.

faceplate graphic is obviously completely different. In signing off UAD’s version, it seems like Eventide has borrowed the faceplate for its own version. But what’s weirder, is that Eventide has held true to its Clockworks layout, opting not to add some of UAD’s special sauce options — even the ones that were on the original hardware unit. The missing bits are the second delay output and an Envelope Follower section; which are both nifty inclusions present on the original unit.

I thought it would be a good chance to check out whether the UAD-2 plug-in would outpace Eventide’s.

The second delay adds a secondary delay line that makes the H910 even more kooky, and you can also blend its signal in to taste. The Envelope Follower is a little different from the original, which tracked an external CV input to alter the pitch. UAD’s version responds to the original signal, altering its own pitch based on its envelope.

At first glance, things weren’t looking that different. If the two companies haven’t shared their IP, there’s definitely been some graphics files floating between the two. This is where it gets a bit weird. After a closer look at the Clockworks-era H910 plug-in, the

The UAD version also comes with a graphic representation of the HK941 keyboard controller you can use to change the pitch. Nevertheless, both plug-ins can be controlled via a MIDI input, and trying to play the graphic keyboard with a mouse isn’t very rewarding.

After going back and forth dialling in the same settings on both plug-ins, it seems to me that both have the same underlying sonic architecture. Even the ‘random’ slippage of the pitch ratio seems to be similarly timed. At the end of the day, if you want that glitchy pitchshifting on your tracks, either of these H910 plug-ins will do the job, and sound fantastic. The only thing that required a higher buffer setting was when I was changing the pitch in realtime via MIDI. That function crapped out my system on a 32-sample buffer, regardless of whether the plug-in was running on DSP or not. Kudos to UAD for adding some more touches to its H910. That said, Eventide has thrown in a H910 Dual Harmonizer into the Anthology X bundle, which stacks two units in parallel and ramps up the maximum delay time to 400ms. UAD Eventide H910: US$249 Eventide Anthology X (includes 17 plug-ins): US$1195

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SOYUZ SU 011 & SU 017

Tube Microphones How do you make a ‘vintage’-sounding mic without cloning an original? Soyuz shows us how. Review: Greg Walker


There are loads of engineers out there seeking the timeless tone of mics like Neumann’s U47 and U67, with one caveat — it has to come at a fraction of the cost of an original. Boutique manufacturers have jumped at the chance to try and fulfil this sizeable gap with more or less accurate clones of various classic designs. With big margins to be had below inflated vintage microphone prices, there’s not much to lose. Brands like Bock, Peluso, Mojave, Wunder, Lauten, Flea, ADK Zigma and Australia’s own Beesneez, even Rode with the K2, have done a wonderful job of bringing great sounding, quality tube microphones to market at comparatively affordable price points. And they’ve all

PRICE SU 017: $5350 SU 011: $1840 SU 011 Matched Pair: $3060 CONTACT Studio Connections: (03) 9416 8097 or sales@

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captured at least some classic vintage tone magic. But has anyone really nailed it? It’s a hard question to answer as it’s nigh on impossible to pin down the exact sound of a 60-year old microphone in the first place. Certainly no one who owns a $12,000 original U47 is going to say a modern mic worth a third of the money sounds just as good. Hang on, actually there is someone who’s saying pretty much that, and his name is Nigel Godrich. FROM RUSSIA WITH LOVE

Mr Godrich’s enthusiastic quotes lead the line in Soyuz’s promotional material and this is no mean feat as he rarely features in the media, preferring to let his production

PROS Wonderful sonic performance with a hint of harmonic colouration Fairly strong proximity effect offers tonal options Fantastic build quality & looks great Easy power status checks via PSU LEDs

CONS Cardboard packing cases No bass roll-off or attenuation onboard Fixed polar patterns Expensive!

SUMMARY The Soyuz microphones back up their good looks with bulletproof engineering and great performance. They excel at all the main applications while delivering tube colouration that is tonally balanced and very musical.

credits (Radiohead, Beck) do the talking. In short, he loves the Soyuz mics. When comparing the large diaphragm condenser SU 017 to his favourite vintage U47, he finds them to be in the same very desirable ballpark. It’s a massive tick of approval for Soyuz. The new Russian company is based in Tula, a town about an hour south of Moscow that lays claim to having the oldest continuously operational microphone factory in Russia (since 1931). Soyuz’s mission statement is simple: build great sounding hand-made tube microphones based on the classic designs of the mid-20th century. Hardly a novel approach in this day and age but when I opened the boxes of the SU 011 and the SU 017 small and large diaphragm models things started looking a little different, and it was hard not to be impressed. For starters, these mics look sensational. Soyuz has abandoned the cookie cutter chassis these clones usually come in for a classy combination of old-school machined brass and cream enamel. The Russian logo lettering lends an extra touch of cold-war retro cool. Both mics exude quality above and beyond what we’re used to seeing from the majority of mic manufacturers these days. Soyuz even ensured the body’s length relative to its width conformed exactly to the golden ratio! Somewhat disappointingly there’s no sign of flight cases. The cardboard shipping boxes contain stylish cream-coloured ‘brick’ PSUs, 5m connecting cables and, in the case of the SU 017, a well-made, cream enamelled suspension mount. Fortunately, the mics themselves nestle in lovely magneticallylocking cherrywood cases with felt interiors and leather padded yokes. They smell nice too. SU 017 LARGE DIAPHRAGM CONDENSER

The Soyuz SU 017 uses the ‘lollipop’ form more commonly identified with Neumann designs like the CMV3 and CMV563. At 226mm long and 55mm wide the Soyuz is a fairly large microphone and weighs in at 950g, requiring a sturdy mic stand. This design allows capsules apart from the standard cardioid (omni and figure 8 are in the pipeline) to be interchanged using the same body. For my review I had a cardioid capsule featuring a hand-tuned, one-inch gold-sputtered diaphragm. This design is based on the dual-element design of the classic Neumann K67 capsule (found in the U67 and U87) and shares that model’s back plate drilling pattern. The tube used in the SU 017 is a 6G1P pentode that roughly equates to an E95F or 6AK5W in the west. Thoughtfully, Soyuz has included an extra tube to delay the pain of sourcing spares internationally. The combination of this Eastern tube, an in-house custom wound cylindrical transformer and a simple point-to-point handwired layout help deliver an SPL rating of 120dB SPL and equivalent noise rating of 20dB (A-weighted). The mic operates at 270Ω impedance and there are no bass roll-off or attenuation switches here — what you see is what you get. Nestled amongst the packaging was an

individualised measurement sheet. On-axis the SU 017 displays a gentle rise between 30Hz and 80Hz, then is pretty smooth up to a gentle 1dB boost between 1kHz and 4kHz with a larger 3dB rise centred at 5kHz. Then another 3dB rise that peaks around 12kHz before steadily tapering off further up. Off-axis response shows a big drop around 62Hz and another big scoop around 1.5kHz before rising back up at around 6kHz. A rather quaint inclusion is the hand signed cards of both the assembler and tester including their photographs (in my case Olga and Vladimir). It’s oddly reassuring to know they both look like kind people. It’s also worth noting that the SU 017 is not a straight clone of any vintage mic. Rather it borrows elements from a number of designs including Neumann’s U67 while mixing in some homegrown ideas of its own.

Soyuz even ensured the body’s length relative to its width conformed exactly to the golden ratio!


With the Soyuz SU 017 up and running on some sessions I soon realised I could rely on it in a wide variety of applications. For a start, the mic performed beautifully on a range of voices. On a fairly quiet male folk singer the extended warmth of the low end and the sweet airy highs combined with some subtle but very nice euphonic colouration in the upper mids delivered a great, ready-made result that sat in the mix effortlessly. On female singersongwriter Lisa Richards, who has a powerful voice with a lot of upper midrange presence, the SU 017 delivered a beautifully nuanced picture of her tone and character without ever getting edgy or harsh. This mic has a pronounced proximity effect and, given the absence of any bass roll-off filter, I found I was positioning singers slightly further away from the capsule (up to three or four feet) than I would with some other mics. Having said that, the results were outstanding and the extra distance also served to minimise vocal artefacts such as popping and sibilance. I never felt the need to use a pop shield or place a de-esser on any of the vocals I recorded using this mic. The SU 017 also worked beautifully on guitars, bringing out the feathery texture of a strummed 12-string acoustic and the earthy thud of a thumbpicked six-string part. On electrics it really shone, lending that lovely tube colouration to bright Fender and thicker Gibson tones. I also used the SU 017 as a mono drum overhead with great results. The mic does begin to saturate at higher volume levels and although the transition is entirely musical and useable, the colouration may not suit every style of music and taste. Finally, I put the SU 017 on my old double bass, about four feet away and aimed halfway between the bridge and the top of the fretboard. The results were sublime; plenty of sweet bottom end but also a nice clarity and airiness in the top end of what can sometimes be an overly dark instrument. Another thing I noticed was how well numerous recordings of the SU 017 sat together. Very little EQ was required on anything and the tonal balance was spot on while being very sonically pleasing. AT 67


The SU 011 is a chip off the old block in many ways. Again the build quality is exemplary and I love the cream and brass look; in particular the ‘prison bar’ brass grille at the head of the cardioid capsule. The cherrywood case has extra storage spaces for alternative capsules (hypercardioid and omni as well as a -10dB attenuator are planned). This pencil condenser comes with a basic clampstyle mic clip in cream enamel and its PSU is more or less identical to its larger brother’s. The SU 011 uses a sub-miniature 6S6B tube made in 1986 that was sourced from ‘dead stock’ courtesy of the former Soviet Union. It has slightly better noise specs than the SU 017 and operates at 120Ω with the same 120dB SPL handling. As far as pencil condensers go it is quite small at 26 x 123mm and weighs only 130g, but it looks the business on a mic stand. The included measurement sheet shows a gentle roll-off below 150Hz and peaks at 4kHz and 13 kHz, before rolling off above 16kHz. PENCIL IT IN

I initially used it on a lovely Martin small body acoustic guitar and the results were sensational, giving me a great balance of body and airy clarity. By virtue of the mic’s proximity effect I was able to dial in the amount of bass required, moving the SU

011 in and out before settling at a distance of about three feet. This sound became the bedrock of an entire album and I was stoked with the result. At one stage during a session I swapped it for my trusty old AKG 451 just to see what the difference would be. The 451 sounded more than adequate but I immediately missed the extra sweetness in the top end and the subtle valve colouration I was getting from the Soyuz. Needless to say I quickly swapped back. The SU 011 worked beautifully on a number of acoustic instruments and I got great tones on six and 12-string guitars, cello as well as electric guitar and percussion. My last test of the SU 011 was on snare drum about four inches above the edge of the rim and the Soyuz was fantastic, delivering plenty of wallop and again that sweetly detailed top end. RUSSIAN REVOLUTION

Overall I was very impressed by the Soyuz microphones. They have the tonal characteristics I look for in quality tube microphones; they’re warm but not dull, detailed without ever getting harsh, but above all, super musical. They look great and are versatile while doing the core jobs such as vocals and guitars with real finesse. There are a few minor gripes such as the lack of flight cases and bass roll-off or attenuation controls that would further increase their utility, but all in all the Soyuz mics are some of the highest quality and

best sounding microphones I have come across in a long time. Unfortunately the effect of the current exchange rate is going to push them well above the budget of many prospective buyers but for those who pull the trigger, the Soyuz sound will provide a lot of joy for a long time to come.

I never felt the need to use a pop shield or place a de-esser on any of the vocals I recorded using this mic

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Built to Create. The PreSonus Studio 192 is the first USB 3.0 audio interface to deliver exceptional sonic fidelity and flexible connectivity in addition to professional monitoring and mixing controls in a single rack-

• Simultaneously stream up to 26 inputs and 32 outputs at 48kHz – or 8 x 14 at 192 kHz. • Take advantage of award-winning StudioLive™ Fat Channel processing on every analog input and the first 8 ADAT inputs with ZERO LATENCY monitoring • Remote control your XMAX™ preamps and create zero latency monitor mixes right from the included Studio One Artist DAW. (3rd-party DAW users can use PreSonus’ advanced UC Surface software to create zero-latency mixes with speaker switching and remote control the Studio 192 preamps via MIDI.)

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AKG C314

Multi-Pattern LDC microphone The C314 — a C414 in a C214 body — but without the identity crisis. Review: Mark Woods

C414 CAPSULE — Inside the grille is the same one-inch, dual-diaphragm capsule as the current C414 XLS and the same on-board suspension mounting. These diaphragms are computer-analysed to deliver high polar-pattern accuracy in single mics and determine which mics are to be sold as matched pairs.

BODY CONTROL — The C314 design is understated when compared to the relatively flashy, lit-up C414, but it still looks like an AKG; silver grille on the front and black grille on the rear. The colours and finish are the same as the new AKG D12VR with a strong steel frame, scratch-proof finish and double-layered metal grilles front and rear.


PATTERN CHANGE — You get physical switches to select from four pattern choices; omni, cardioid, super-cardioid or figure-8. On one side of the mic body a switch enables a simple 100Hz, 12dB/ octave high-pass filter.

PRICE $1499 CONTACT Hills SVL: (02) 9479 0830 or www.

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PROS C414 sound quality 8dBA self-noise Multi-pattern Rugged build & clip indicator


SUMMARY The C314 isn’t quite the C414, but it’s got darn well everything you need onboard — a C414 XLS capsule, multiple polar patterns, pad and clip indicator. It’s in a body that won’t complain about being outside, and its off-axis response delivers neutrality in the studio and plenty of gain when used live.

SHOCK BARRIER — The supplied spider-style shockmount is made from plastic so its light, but it feels strong and well-balanced with the mic in place. I use these shockmounts for live shows and like the way they form a little barrier around the mic, like a reverse proximity effect for over eager performers. A regular hard standmount is also included and it all gets delivered in a good case with a custom slip-on wind shield/pop filter.

I’ve been playing with AKG’s C314 for a while now, and I can’t help but wonder: what is AKG thinking? The C314 just seems too good to be true. Especially for someone who has a hankering for a C414 but without the capital to match. The AKG C414 is a classic. Since the ’70s it’s been put in front of anything, sometimes everything, with good results. Classical people have always liked them because they’re quiet and transparent; rock people for the way they record acoustic guitars, drum overheads and vocals. Live engineers like their compact size, robust build and resistance to feedback. AKG updates the C414 every 10 years or so, and sure there’s been variations in sound with different models and manufacturing materials but it’s always maintained that signature smooth midrange and distinctive, airy high end. The current versions are very nice. The silver-faced C414 XLS and the slightly brighter-sounding, gold-faced C414 XLII, are luxuriously engineered with nine polar patterns, multiple pad settings, multiple high-pass options, and electronic switches with indicator LEDs. Good value they may be, but the price for these new C414s is quite high and perhaps beyond many project studios. I also think they might be too nice for the majority of live shows. At a certain level of production, sure, but in the real world it’s a lot of money’s worth of microphone at risk in unpredictable live environments. Stage

It’s so uncoloured below 1kHz you don’t really notice the mic at all unless the source is close enough to induce some proximity effect

mics get bumped and occasionally knocked over — regularly during band changeovers at folk festivals, I’ve found — and it might rain. AKG has gone some way to addressing both these issues with its latest release: the AKG C314. It combines the capsule from the C414 XLS with the body from the single-diaphragm, fixed-pattern C214 to create a product some users, me included, see as combination of the best elements of both.

when I was setting up for a recording. It’s bright without being forced or spiky. I used it for vocals on a session that was already half done and it made the mic I’d been using sound grainy… hate that. I then started using it on everything I could for a while and it doesn’t miss technically, all recorded sounds are usable and I found it really brought out the character of different preamps.


A tougher test was the annual Guildford Banjo Jamboree. Don’t laugh, it’s one of my favourite events and an excellent test gig for microphones. All acoustic acts, with not a DI in sight. The festival takes place in three venues over three days. It starts in a small wooden music hall built in 1853, moves to a typically reverberant town hall the next day, and finishes with an outdoor concert on the final day. This year the C314 did all the single-mic acts and got used on a variety of instruments for the multi-mic acts. I enjoyed it from the start. Live PAs, large-diaphragm condenser mics, delicate instrument sounds and performers who are often way off-mic is not a combination for the faint-hearted sound mixer… but it can deliver the purest sound you’ll ever hear through a PA. Gain is the aim and an accurate off-axis response affects how far the mic can be opened up before feedback. By this measure, the C314 was as good as any

In the studio, my first impression was that it’s almost eerily quiet, and the off-axis response is natural. Its combination of high sensitivity (20mV/Pa) and low self-noise (8dBA) contributes to the C314’s very modern sound. It’s also accurate and fast, with great transient detail and a slightly flattering overall response. On paper it’s close to flat in the Omni setting, but compared to the C414 it has a wider and deeper dip centred around 2kHz, and a wider boost above 5kHz, in the directional settings. Through speakers it’s so uncoloured below 1kHz you don’t really notice the mic at all unless the source is close enough to induce some proximity effect. This is not an aggressive mic; the couple of dB attenuation around 2kHz contributes to the overall smoothness and tames harshness in strident vocals. The high-end boost delivers the distinctive AKG airy sheen to vocals and it won an acoustic guitar shootout by having the shiniest top end


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condenser mic I’ve used, delivering easy volume in each of the different environments. It’s beautifully tuned for live sound too; the vocals are rich and engaging, it doesn’t get harsh at high levels and there’s plenty of sparkling detail from the instruments. It’s transparent and does that thing very good microphones do where they seem to be able to reach back to instruments, even if they’re a metre or more away, and bring them in closer to the microphone. I’ve written before about how the audience at these acoustic/Americana events take the sound very seriously and this year I lost track of the number of people asking what mic I was using… and none of them were asking because of its looks. THE CONVINCER

I’m convinced. This is a great mic, a C414 in a C214 body. For AKG it fills a gap in the range between the top-selling, entry-level C214 and the fully-pro C414s. Also available in computer-matched pairs, the C314 will appeal to users who want the best possible sound quality and are happy to save a few dollars by forgoing a few patterns and features. It would make itself right at home in any studio and the simpler body makes it more suitable for live work than its more expensive sibling. Made in Vienna, it has a three-year warranty but like all AKG microphones you’d expect it to be around for the long haul. I want one.

PAD OVERLOAD — On the other side of the mic is a single 20dB pad with an overload light that changes from green to red when the input level is high enough to need the pad engaged. It either flashes or stays on for a few seconds, depending on the volume. It’s a great idea, as engaging pads unnecessarily reduces a mic’s dynamic range, but it’s often hard to tell whether you need the pad to prevent the mic’s internal electronics overloading. The light is on the side of the body below the switch, so you have to position the mic in you can see the LED while you blast it with the full force of your loud sound to see if it flashes. Aside from the comfort of knowing you’re protected, I found it reassuring the C314 took a lot of level to set the clip indicator off. It made me confident the pad would only be needed for extremely loud, bass-heavy sources; inside a kick drum for instance.









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Wireless Microphone System The less you mess with Sennheiser’s AVX System, the better. Review: Preshan John


There’s an endless number of ways to bring a cinema shot alive and the tools to do it — sliders, stabilisers, jibs, drones and dollies — are now accessible to even amateur cinematographers. You’ll rarely see a wedding or event videographer without some mechanical contraption clogging up the aisles these days. Recording incidental sound for film, however, — like dialogue or ambience — is generally more systematic than it is creative. Primarily, it’s got to be present, intelligible, natural and unobtrusive. The creative stuff is mostly left to sound design and post-production.

PRICE Handheld Set: $1499 Lavalier Set: $1499 Lavalier Pro Set: $1749

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CONTACT Sennheiser Australia: (02) 9910 6700 or

So when it comes to camera audio, especially run ’n’ gun DSLR video stuff where you’ve donned both cameraman and sound recordist hats, the sound usually serves the vision. It makes the typical requirements for a wireless system — sets up quickly, is easy to use, and performs reliably — even more pertinent. I reviewed the Audio-Technica System 10 wireless camera mic system a couple of issues back. Even after extensive use, the System 10 continues to be a great example of no-frills yet no-fuss functionality — a valuable trait when your shooting is largely impromptu and without

PROS Quick auto setup & pairing Tiny, but well-made & designed Auto gain-riding works well Good sound quality

CONS Headphone output on receiver would be nice

the luxury of much setup time. In a similar ‘no fuss’ vein, Sennheiser has sprouted the AVX wireless camera mic system. Unlike the System 10, this doesn’t have an ‘entrylevel’ vibe about it. With a tag-line of ‘Relax, it’s an AVX’, Sennheiser has added the bells and whistles behind the scenes, so you have to do less fiddling on the field. The specs are all there for a good run ’n’ gun option: 24-bit/48k sampling, 10-hour transmitter battery life, five-hour receiver battery life, AES 256-bit transmission encryption, and lithium ion battery. The system

SUMMARY The Sennheiser AVX wireless system is easy to use, built extremely well, performs reliably and doesn’t get in the way of your creative flow. Sound quality is impressive, especially from the handheld AVX-835 microphone. It’s not the cheapest camera audio accessory, but if you want to evade audio distractions while filming, the AVX should rate high on your list.

is offered in three combinations, all of which include the receiver: the Handheld Set, which comes with the AVX-835 handheld supercardioid mic; the Lavalier Set with the AVX-ME2 lapel mic and bodypack transmitter; and the Lavalier Pro Set, which includes the better-quality AVX-MKE2 lapel mic. We were sent the receiver with both the AVX-835 and ME2 mics for review. ROADWORTHY

Sennheiser hasn’t skimped on build quality. Though they’re diminutive, every piece of kit in the AVX package feels like it could handle being strapped to a warzone journalist’s kit. The ingenious micro receiver is a welcome progression from typical palm-sized on-camera receivers. For camcorder/broadcast-type cameras, it will neatly plug straight into an XLR input — no cables involved. What’s more, turn on your camera’s phantom power and the receiver will automatically switch on and off with it. If you have a DSLR that only accepts audio via jack input, the kit comes with a female XLR to 3.5mm adapter cable. The receiver can rotate within the hotshoe mounting clip for flexible camera-top positioning. Alternatively, a belt clip can slot into the hot-shoe on the receiver to wear it on your hip, or clip it to a bag. Sennheiser seems to have thought of everything. Well, almost everything. The one thing I missed on the AVX receiver was a headphone output for monitoring the microphone — something I grew accustomed to when using the System 10.

Preferably you want to monitor at the recording end anyway, to ensure you’re not clipping, but it can come in handy when using cameras that don’t have their own. Charging the units only takes a couple of hours, but allow some extra time for the initial power up. All AVX units use a mini USB plug, so you’re never short of ways to get some juice into them. Both the handheld microphone and the belt pack transmitter display pairing status and battery life readouts on their screens. You can check the battery life of the receiver on the four green lights on its side, which illuminate by pressing the power button. LICENSE TO NIL

The AVX wireless system operates in the licensefree 1.9GHz DECT range. Cordless phones are the main devices using the band, so unless you’re standing in the middle of a call centre, finding an interference-free channel is typically hassle free. Up to 12 AVX systems can run simultaneously without stepping on each other’s toes. It’s a pretty brainy system according to Sennheiser, with a lot going on under the skin. AVX will dynamically choose unhindered frequencies to operate within while covertly scouting for clear channels. The first sign of interference prompts an inaudible, automatic switch over to another clean channel. There is a 19ms latency penalty for all the dancing around required in this range, but that’s not a big problem if the talent isn’t monitoring their signal. Transmitting power is up to 250mW and adaptive to distance. It won’t blast out its maximum RF

if the receiver’s only a foot away, only what’s required to maintain a secure connection, which in turn conserves power. AVX also has a built-in gain-rider called Dynamic Range [thanks Sennheiser for co-opting a commonly used phrase – Ed]. The feature is designed to match a camera’s input sensitivity and keep signal levels within an optimum range to prevent clipping or sounds that are too quiet. Being an audio guy, I’d usually prefer to have total control over levels myself, without any ‘intelligent’ intervention. But when using the AVX, I found Dynamic Range actually resulted in less stress for those time-restricted point-and-shoot moments — and the more I used it, the more it gained my trust. It’s actually a relief if you’re changing input levels on a DSLR, which typically requires a number of button-presses. TAKE IT TO THE STREETS

The first time I used the AVX system for video recording was for some informal, outdoor piece-tocamera promo videos — a perfect opportunity to put the two microphones’ intelligibility and wind rejection to the test. Sennheiser’s big promise is that the system should be ready to go in seconds. This was true in my experience. It was simply a matter of pressing the Pair button on both the transmitter and receiver, and waiting for the flashing green light to turn solid. From then on, the receiver automatically paired with either transmitter when powered up. The specified range of the AVX system is 30m for regular interview-type shooting. In my tests it AT 75

managed close to 50m line-of-sight, but it dropped out the moment I turned a corner. Perhaps not the system for those Getaway ‘where is the presenter? Oh, there she is, on the boat’ long zoom-ins. The handheld mic performed very well, although it was slightly prone to plosives. It sounds fantastic though, delivering polished, natural speech reproduction and an extended high end. The super cardioid capsule has good off-axis rejection, so it’s comfortable in relatively noisy environments provided you keep it close to the sound source — which is just as well because I thought its proximity effect was rather pleasing on spoken word. The red/ green status indicator light doubles as the On/Off switch, and its position makes accidental power cycles a near impossibility. Dynamic Range, the auto level control feature, did a great job with the AVX-835 mic. I gained up my mic input about a third of the way, paired the handheld mic with the receiver and let AVX do the rest. It managed ordinary fluctuations in SPL with commendable finesse — not once did the camera’s input clip or drop off to an unusably low level. I should add that changes in level aren’t always drastic. Don’t expect results holding the mic six feet from the talent’s mouth in the hope it’ll sufficiently crank up the gain. In other words, keep your audio wits about you when filming, and you’ll be pleasantly surprised when AVX helps you out. It was a slightly different story with the MKE2


Keep your audio wits about you when filming, and you’ll be pleasantly surprised when AVX helps you out

lapel mic. The AVX system’s automatic level feature wasn’t as nimble with the lapel mic’s input. Sonically, the ME2 capsule sounded pretty good — a little ‘congested’ when compared with the airiness of the handheld, but that’s to be expected with any body-worn mic. It pays not to clip the mic right up under the talent’s neck, or you’ll end up with more honk than an angry truck driver — I found clipping it about 15cm down from the collar provided natural-sounding results. If you’ve got someone bellowing into the lapel, you can reduce the output further by pressing the AF Out button on the receiver, which attenuates the signal by four 10dB increments.

The AVX system proved to be very reliable over multiple uses, with all units demonstrating an impressive battery life. I gravitated toward the AVX-835 handheld mic for most applications because it had that slight edge in sound quality. It worked best for solo presenters. Catching interview dialogue can be a bit harder with the tight pattern, and for anything else, there’s not much chance keeping the big black mic out of frame. It’s not to say the ME2 was a poor performer — it still sounds great on spoken word and is far less conspicuous than the AVX-835. The ME2 was better suited to a calm and collected approach, where you have the time to carefully position it, check levels, and ask the talent to run the cable down their shirt. And remember you can buy an AVX set with the better quality MKE2 capsule if lapel mics are your thing. Sennheiser’s AVX system does what it says it will. More importantly, there’s a payoff to all those behind the scenes features; a distraction and inconvenience-free experience while you’re trying to nail that perfect shot. In my experience, it really was the closest to a set-and-forget system I’ve used. If you’re willing to pay a little extra for a little less hassle, then the AVX system might just be the perfect fit for you.

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