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Editor Mark Davie mark@audiotechnology.com.au Publisher Philip Spencer philip@alchemedia.com.au Editorial Director Christopher Holder chris@audiotechnology.com.au Publication Director Stewart Woodhill stewart@alchemedia.com.au Art Direction & Design Dominic Carey dominic@alchemedia.com.au
Regular Contributors Martin Walker Michael Stavrou Paul Tingen Graeme Hague Guy Harrison Greg Walker James Roche Greg Simmons Tom Flint Robin Gist Blair Joscelyne Mark Woods Andrew Bencina Jason Fernandez Brent Heber Gareth Stuckey
Additional Design Daniel Howard daniel@alchemedia.com.au Advertising Philip Spencer philip@alchemedia.com.au Accounts Jaedd Asthana jaedd@alchemedia.com.au Subscriptions Miriam Mulcahy mim@alchemedia.com.au
E: info@alchemedia.com.au W: www.audiotechnology.com.au
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 2015 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. 10/6/2015.
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Studio Microphones MACKIE MCU-PRO 9 Alps touch-sensitive faders, a full-sized backlit LCD and V-Pots for fast tweaking – the ultimate in hands-on command. AT 4
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TURRAMURRA PROFESSIONAL STUDIO DIVISION 1263 PACIFIC HIGHWAY, TURRAMURRA, NSW. TEL: (02) 9449 8487 FAX: (02) 9449 3293 WEB: www.turramusic.com.au EMAIL: hitech_sales@turramusic.com.au
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COVER STORY
Sia: 1000 Forms of Fear Producer Greg Kurstin 16 has nothing to fear… only more gongs
Be Afraid: Alien: Isolation’s sound design team dial in the audio terror
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Calling All Hipsters: Sugar Mountain Festival Live
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Studio Focus: Phoenix Sound Studios
Emergence AS8 Pro S Monitor System
Rode Stereo Videomic X
50
Last Word: with Bill Putnam Jnr, Founder Universal Audio AT 6
ISSUE 22 CONTENTS
42
AEA Nuvo N22 Ribbon Microphone
Mackie DL32R Tablet Rack Mixer
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28 38
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Introducing the new NTR active ribbon microphone from RĂ˜DE The finest ribbon microphone ever made. Hear it for yourself at rode.com/ntr
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GENERAL NEWS
SE RUPERT TUBE MIC sE Electronics has once again teamed up with Rupert Neve to produce the RNT large-diaphragm tube microphone. The mic features a three-position high-pass filter and nine-step polar pattern dial going from figure-eight to omni and everything in between. It has a high-voltage tube circuit within the microphone itself and Rupert Neve’s input is evident with the floor box containing a discrete, Class A amplifier circuit using the same custom op-amps as Rupert Neve Designs’ 5088 analogue
mixing console. There are also two of Rupert’s custom transformers and a custom mic capsule designed by sE to provide extreme headroom (153dB) and high-bandwidth frequency response. The ample headroom means there’s no need for a pad, but the amplifier circuit in the floor box has three gain settings of -12dB, 0dB (flat), and +12dB. Sound & Music: (03) 9555 8081 or www.sound-music.com
AKG C CHANGE The AKG C314 is a new dual-diaphragm condenser microphone featuring low self-noise, high dynamic range, and a neutral sound. According to AKG the C214 is the best selling recording mic in the US, so little wonder it wanted to fill the gap between it and the legendary C414 (and its modern variants). Key features include: four selectable polar patterns; the same C414 XLS gold-plated membrane for “high linearity and neutral sound”; overload LED Detection Display, which indicates ‘too loud’ SPLs; and the
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usual array of accessories. Also of note is the integrated capsule suspension, reducing mechanical noise and resonances. A double-mesh, all-metal grille protects the capsule and ensures high RF immunity without affecting the microphone’s acoustical performance. A 20dB attenuation pad and bass-cut filter is included for close-up recording and reduction of proximity effect. Hills SVL: (02) 9749 0830 or www.hillssvl.com.au
NOVATION HAS DISCO BALL Disco legend, synth nut and renaissance man, Giorgio Moroder, finally has his own synthesiser. Introducing the limited run (500 units) MoroderNova. Giorgio was a fan of Novation’s MiniNova so it wasn’t too much of a stretch to produce a special edition, with the trademark Moroder ‘mo and shades’ motif. Yes, there are plenty of signature Moroder sounds: from the timbres of
his disco classics with bands such as Sparks and Donna Summer, to the incredible sounds of the Top Gun and Scarface soundtracks, Novation’s sound designers have gone to great lengths to match each with the original as best they can. Innovative Music: (03) 9540 0658 or sales@innovativemusic.com.au
SENNOGEE SYNERGY Apogee and Sennheiser have combined to produce a couple of iOS recording systems: the ClipMic digital and the MKE2 digital. Both Sennheiser mics are particularly well-suited to recording speech, designed with the journalist, lecturer or podcaster in mind. Apogee has taken care of the A/D conversion side of things, with digital resolution up to 24-bit/96k. The mics plug in to the Lightning connector on any iOS device and the headset socket can be used to monitor the audio being recorded. Apogee has also created a recording app for iOS devices called MetaRecorder.
It offers multi-take recording, tagging and file organisation for any field recording scenario and complete software control of the mic input level and recording presets. MetaRecorder has been optimised for recording film audio, with the possibility of applying metadata tags to audio files, then sharing the recorded audio and Final Cut Pro XML files to Dropbox directly from MetaRecorder. You can even control the app using an Apple Watch! Sennheiser: www.sennheiser.com.au
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GENERAL NEWS
INTERFACES & CONVERTERS OF ALL PERSUASIONS.
$1549 National Audio Systems: 1800 441 440 or sales@nationalaudio.com.au
~$3000 Audio Chocolate: (03) 9813 5877 or www.audiochocolate.com.au
Innovative Music: (03) 9540 0658 or www.innovativemusic.com.au
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More than a USB 3.0 audio interface, the Presonus Studio 192 has some nifty console ‘centre section’ features. The new 26 x 32 interface records at up to 192k and combines eight digitally controlled XMax Class A solid-state mic preamps and Burr-Brown converters (for 118dB of dynamic range) with StudioLive Fat Channel signal processing. The ‘command centre’ features are in its ability to manage speaker switching and talkback (with onboard condenser microphone), featuring main mix mute, mono, and dim. The remote controllable preamps add a separate digital volume-control circuit for digital recall. The Studio 192 I/O includes: two front-panel mic/instrument inputs and six rear-panel mic/ line inputs; 16-channel ADAT optical in and out (eight channels at 88.1 or 96k); coaxial, stereo S/PDIF I/O; and BNC wordclock I/O. You get eight balanced TRS outputs, balanced stereo main outputs, and two headphone amps with independent outputs and level controls. Introducing the MP8d eight-channel mic preamp with A/D conversion – Antelope Audio’s newest addition to its lineup of mic pres. The MP8d features eight channels of Class A mic preamplification, with the preamps taking their inspiration from Antelope’s recentlyreleased MP32 and Zen Studio interface. [BurrBrown chips? - Ed] In addition, the MP8d also sports A/D conversion for each channel with its Acoustically Focused Clocking jitter management algorithm and Oven-Controlled crystal oscillator, powered by a proprietary PSU. Apart from the preamps there are inserts for connecting outboard gear available, as is a headphone amp output. The analogue connectivity comes via a D-Sub 25 connector directly outputting the mic signals at line level. There’s plenty of digital connectivity, with S/ PDIF, AES/EBU, TosLink ADAT and MADI. The MP8d also allows for 24-bit/192k streaming of 24 simultaneous I/O via USB, utilising Antelope’s custom-built PC/Mac drivers. Not just another eight-wide rackmount preamp. Aside from the eight console mic pres and two discrete JFET instrument inputs, ASP800 also has two ‘Retro Channels’ featuring tone controls HMX and Iron. HMX is a
custom harmonic distortion designed to emulate the sweet, thick tones often associated with 1960s tube designs, while Audient’s brand new British transformer saturator, Iron adds sparkle, replicating the transformer ‘zing’ of the British console sound of the 1970s. Both controls add interesting new colours to the sonic palette. Key features also include 118dB Burr-Brown A/D converters, ADAT connectivity, a wordclock input, a -15dB pad per channel and balanced analogue line outputs. Audient’s iD14 interface provides a pair of Audient console mic pres, Burr Brown converter technology, console-style monitor control, JFET DI and something called ScrollControl. As well as offering you two mic pres, iD14 has a JFET DI input. Designed to replicate the input stage of a classic valve amplifier, you can plug your guitar, bass, keyboard or drum machine straight in and start recording instantly. Then there’s ScrollControl – a feature that turns iD14’s volume encoder into a virtual scroll wheel that allows you to adjust DAW hosts, plug-in parameters, iD14’s mixer app and even scroll through your iTunes library. When not using the virtual scroll wheel, the iD button can be assigned to control iD14’s monitoring features, including dim, cut, polarity reverse, mono sum, talkback and cue mix monitoring. The original release of the RME Babyface, with its small form factor packed with I/O, redefined what many expected from a desktop interface. RME’s new Babyface Pro is a dramatically face-lifted USB 2.0 interface with many of the RME features we’ve come to expect: a large, multifunction dial, input and output meters display, two instrument/mic inputs directly accessible on the unit, a pair of XLR monitor outputs and 3.5mm/6.5mm headphone outputs. The two digitally-controlled preamps have a gain range of 71dB, adjustable in 3dB steps. Digital I/O comes in the form of an optical TOSLink port, which you can use either with ADAT or S/PDIF devices. MIDI connectivity is available via an included breakout cable. TotalMix FX software comes as standard, allowing extensive mixing/routing capabilities when tracking or monitoring. But even when it’s not hooked up to a computer, the Babyface Pro allows routing and mixing of input to outputs directly on the device in stand-alone mode.
MORE NEWS AT www.audiotechnology.com.au
iD14: $449 Innovative Music: (03) 9540 0658 or www.innovativemusic.com.au
$1299 Innovative Music: (03) 9540 0658 or www.innovativemusic.com.au
From $3299 (Apollo 8 Duo) CMI: (03) 9315 2244 or www.cmi.com.au
Universal Audio has announced its nextgeneration line of Apollo audio interfaces. The A/D and D/A conversion has been improved; 2dB better dynamic range on the A/D side, and an 8dB improvement on the D/A. All feature new monitoring functions and built-in Thunderbolt (instead of it being optional), with the ability to cascade up to four interfaces. There’s still a standard Firewire Apollo available, though you sacrifice a little in the latency stakes. As well as the revitalised Apollo and Apollo 16 ($4999) interfaces, there’s the all-new Apollo 8p ($4999), which comes with eight of UAD’s Unison-enabled preamps for tracking through classic preamp emulations from Neve, API, and Universal Audio. You can, of course, record through more plug-ins from Lexicon, Studer, Manley, Ampex and the like using the onboard UAD-2 DSP with an incredibly low 1.1ms roundtrip latency at 96k. The new interfaces come bundled with the ‘Realtime Analog Classics Plus’ UAD plug-in bundle. But most importantly, they come in a new blacked out look. The Black Lion Audio Micro Clock MkIII is a redesign of its MkII predecessor. Core technological improvements include lowerjitter crystal oscillators, higher-quality galvanic transformer isolation in the signal path, dedicated output drivers with better isolation, higherprecision signal division, and much heavier noise filtration throughout the circuit. All of this results in a more robust, lower-jitter clock signal than the Micro Clock MkII, with even more ‘desirable harmonics in the clock’s spectral band’ that ‘impart a musical quality’ to the conversion process. The feature set has expanded to include an LED frequency display, six BNC outputs capable of up to 384k, AES and RCA S/PDIF outputs capable of up to 192k, an optical S/PDIF output capable of up to 96k, and removable 1U rack ears.
Remote-Controlled Digital Mixer
Any device. Anywhere. The freedom to mix. Ui packs a complete digital mixing system into a compact, road-rugged stagebox format, with built-in Wi-Fi that lets you mix remotely from any device direct from your browser, supercharging your live sound with processing from some of the biggest names in the business - dbx®, Lexicon® and Digitech®.
Network Audio Solutions: 1300 30 66 70 or www.networkaudio.com.au
Distributed by
40 Kent Road • Mascot NSW 2020 Ph: (02) 9582 0909 • Fax: (02) 9582 0999 www.jands.com.au • info@jands.com.au
MORE NEWS AT www.audiotechnology.com.au
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LIVE NEWS
MIXING CONSOLE BIG GUNS ROLL OUT THEIR BIG GUNS… THEIR LITTLE GUNS & SOMETIMES BOTH
Group Technologies: (03) 9354 9133 or www.gtaust.com
Jands: (02) 9582 0909 or www.jands.com.au
Price: From $3999 Yamaha: au.yamaha.com
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Digico has the S21 pegged as its gamechanging, disruptive, worldwide shakeup machine, and there’s no question the sub $10k digital mixer market won’t be the same. Harnessing Digico’s use of the FPGA and its mature software, the S21 packs: 24 mic/line inputs, 12 analogue outputs, 96k operation, integrated USB 2.0 audio I/O, customisable bank and channel layout, two ethernet connections, two DMI expansion slots (up to 64 I/O per slot), wordclock, and two AES I/O (mono). The software features include: 46 buses, user definable macros, a gate and compressor on each channel and bus, four assignable multiband compressors, four assignable DigiTubes, eight FX engines, 16 assignable 32-band graphic EQs, 16 flexi buses mono/stereo, 40 flexichannels mono/stereo. Soundcraft’s Vi series has won plenty of friends over the years for its operator-friendly usability. There isn’t a person in the industry who doesn’t love the Vistonics UI. New to the Vi range are the Vi5000 and Vi7000 digital consoles, which replace the Vi4 and Vi6. The consoles match a compact control surface with a new Local Rack and Active Breakout box hardware. There’s more than enough DSP to throw at Lexicon effects and BSS processing. Another cool feature on the Vi is the ability to monitor the status of any compatible Shure or AKG radio mic directly from the console surface (currently that is Shure’s ULX-D and QLX-D systems, and AKG’s DMS800 and WMS4500 systems). The engineer is provided with real-time visual displays of battery life, RF status, mic muting and internal clipping right from the Vistonics screen, avoiding the need for constant over-the-shoulder looks to check the mic receivers — brilliant! We’ve seen the CL, QL, and more recently the Rivage PM10 flagship. Now Yamaha closes the loop on its digital mixing range with the sub $6k TF series. The TF5, TF3, and TF1 respectively feature 33, 25, or 17 motorised faders along with 32, 24, or 16 rear panel analogue inputs featuring recallable Yamaha D-Pre preamps (for the first time in a Yammy digital
console). TF stands for ‘TouchFlow’ which refers to Yamaha’s touchscreen-based UI. There’s some nifty stuff like one-knob Comp and EQ control, there’s also an innovative GainFinder feature which has a smart crack at setting optimum gain for individual input signals. Just when Avid’s Venue live consoles have been getting a bit long in the tooth, first we see the S3L-X and now the large-format, flagship S6L. There’s plenty of processing — with over 300 processing channels — thanks to a chunky engine, combined with a very modern touchscreenbased UI. Highlights include: high-performance preamps running at high sample rates; integrated, on-board plug-ins from Avid and its development partners; Pro Tools capabilities, including Virtual Soundcheck, over Ethernet AVB and Thunderbolt with no interface required; share I/O across multiple networked systems with advanced gain tracking; and instant access to the most important functions through hi-res touchscreens and OLED displays. A complete Venue S6L System includes a control surface, engine, and I/O components. New V3 software introduces over 40 new software and hardware features and updates for the SSL Live console range . The L500 becomes the L500 Plus with v3 now providing 256 mix paths (up from 192) with a doubling of effects processing power (depending on the effects selected). The L300 also enjoys a power increase from 128 to 192 mix paths. Version 3 also includes a new Super-Q feature, an optional Dante interface card, Remote Control software, Console Expander Mode, User Interface changes, new Effects, enhancements to the Solo system and a collection of features designed for customers who broadcast their productions. SSL Live V3 is due for public release in May and will become standard on all new consoles. Traditionalists might well be shaking their heads: what’s the best accessory for the latest rackmount digital mixer? Knobs and faders. Introducing the StudioLive CS18AI mix controller from PreSonus.
MORE NEWS AT www.audiotechnology.com.au
Avid: www.avid.com
Amber Technology: 1800 251 367 or professional@ambertech.com.au
Price: $3699 National Audio Systems: 1800 441 440 or sales@nationalaudio.com.au
Price: $369 – $1999 CMI: (03) 9315 2244 or www.cmi.com.au
Price: from $8999 Technical Audio Group: (02) 9519 0900 or info@tag.com
It’s a control surface for StudioLive RM-series rackmount, Active Integration digital mixers and PreSonus’ Studio One DAW. Networking with StudioLive RM32AI and RM16AI mixers via AVB Ethernet and PreSonus UCNET technology, the StudioLive CS18AI offers touch-sensitive motorised faders (a first for Presonus’ mixers) and enables complete hardware control of all mixer features, managing up to 64 channels. The new controller also makes it even easier to do virtual soundcheck, live recording, and studio mixdown with PreSonus’ Studio One DAW and Capture recording software. Meanwhile, there’s still life in compact analogue mixing, with Mackie launching two new models in the V2 series of its ProFX range. The V2 series has been expanded with two new models, the four-channel ProFX4v2 and the 30-channel ProFX30v2. The biggest drawcard is Mackie’s new Class-A Vita preamps (“virtually noiseless, even at high gain levels”), along with a new ReadyFX effects engine — each mixer features a choice of 16 reverbs, choruses and delays. There are six models available, covering applications ranging from four to 30 inputs, you’re sure to find one the right size for your needs. All mixers in the series except the ProFX4v2 also feature built-in USB recording and playback. Allen & Heath has launched new ‘Chrome Edition’ models of its GLD mixers, with new features and FX. GLD Chrome firmware adds Auto Mic Mixing capability and further additions to the processing suite. The automatic mic mixer (AMM) can be configured to work across all 44 microphone sources (none of the usual restrictions of a 16-channel insert-based system). The new ‘Deep’ plug-in architecture allows users to select from a number of different processing units on every input and mix channel, such as the two new RMS/VCA-inspired compressor models. The new version 1.5 Chrome firmware also features several additions to Allen & Heath’s on-board FX suite, including a new Stereo Tap Delay, Bucket Brigade solid state delay emulator, and Echo, the classic tape echo system.
BOLT OUT OF THE BLUE Supercharge your desktop DAW system Ultra-low latency Lightning quick 10Gb/s Thunderbolt transfer rate 2 x great-sounding preamps Full metering Auto gain feature Pro-grade converters
Zoom TAC-2 Thunderbolt Interface
ZoomAustralia
MORE NEWS AT www.audiotechnology.com.au Dynamic_Zoom-TAC_AT104.indd 1
dynamicmusic.com.au
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15/09/14 3:54 PM
STUDIO FOCUS:
PHOENIX SOUND STUDIOS When Justin Humphries was choosing a name for his new studio, there was really only one choice. After losing his home and studio in the Black Saturday fires, then putting in years of hard work to resurrect something from the ashes, Phoenix Sound Studios just felt right. “It’s taken a lot of hard work and determination over the past six years to recover and rebuild my life after losing absolutely everything,” said Justin Humphries. “I’m happy to say I now have Phoenix Sound Studios to show for it.” Located in Hurstbridge, north of Melbourne, the facility comprises a large 10m by 7m tracking room, a generous control room of the same dimensions, and a smaller vocal/drum booth. Humphries also built in two amp isolation booths. All the rooms are visible to each other, and able to comfortably accommodate large bands and small chamber orchestral groups. Humphries made sure to incorporate plenty of natural light and not obscure the beautiful scenery. Remote computer control and 16-channel individual personal monitoring rounds out the interconnected atmosphere of Phoenix, giving it a great creative flow. As well as engineering, Justin has been a session drummer since the late ’80s, and established an online drum tracks service that had a global client list. In the main tracking room are two Yamaha custom drum kits, with Sabian cymbals, as well as Fender bass and guitar amps, keyboards and various percussion pieces. At the hub of the control room is a Tascam DM4800 64-channel mixing console and Tascam 24-bit/96k conversion. Outboard gear includes multiple Avalon VT-737SP black anniversary edition channel strips, a Universal Audio 6176 channel strip, and a Bellari tube compressor. The mic collection includes a Neumann U87ai, Rode K2, and a matched pair of AKG C414s. And Humphries has a pair of Event Opal monitors to sit alongside his trusty Yamaha NS10s, and KRK Expose E8B studio monitors. Phoenix Sound Studios is currently working on Melbourne guitarist Nick Freer’s album featuring the talents of top Australian musicians Andrew Gander, Phil Turcio, Carlo Barbaro and Matt Clohesy. As well as music production, Phoenix Sound Studios also offers restoration and forensic audio services.
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A V A I L A B L E
N O W !
A ground-breaking approach to compact live sound Yamaha TF Series digital mixing consoles take live sound reinforcement to a new level of refinement. TouchFlow Operation™ makes it easy to achieve a perfect mix, while recallable D-PRE™ microphone preamps contribute to superior sound. Advanced processing, live recording capability, and operation with high-performance I/O racks makes them an outstanding choice for a wide range of applications.
Touchflow Operation™ intuitive user interface optimised for touch panel control
Applications for iOS/Mac/Win that work seamlessly with the console
FIND A DEALER au.yamaha.com
EVENTS AND PROMOTIONS yamahabackstage.com.au
LIKE US ON FACEBOOK facebook.com/yamahabackstagepass AT 15
FEATURE
The harder Sia tries to be anonymous, the more her talent is in demand. Luckily, producer Greg Kurstin has the chops to keep up. Story: Paul Tingen Photos: Lindsey Byrnes
Artist: Sia Album: 1000 Forms of Fear AT 16
“Sia comes in like a whirlwind,” says songwriter and producer Greg Kurstin, explaining what it’s like when the Adelaide songstress shows up at his studio in Los Angeles. “We tend to write at least a song a day, often more. She is a writing machine! Most of the time what we do is very spontaneous, and very fast. We both like to be decisive. She’s one of the fastest people I’ve ever worked with. She doesn’t only write very quickly, she also records really fast — often doing just one take — so you have to be sure you’re in record!” Kurstin was speaking in the days leading up to the 57th Grammy Award ceremony, where Sia’s smash hit song Chandelier was nominated for a whopping four Grammies, including the prestigious Song of the Year and Record of the Year awards — the latter also includes a nomination for Kurstin, as the song’s producer and engineer. The raw lyrics to Chandelier — 1, 2, 3, drink, Throw ‘em back, ’til I lose count — reflect that Sia’s current success has a triumphant ‘against all odds’ element. Over the years her career has been slowed down and nearly derailed by a number of personal issues; ranging from a schizophrenic father, a boyfriend’s death in 1997, extreme shyness, depression, alcohol and drug addiction, ill-health, sexual identity issues, stage fright and a general aversion to being famous. All this led her to come close to committing suicide in 2010, and today she often performs with her back to the audience, hiding her face — normally the focus of any pop singer’s brand — and pushing her trademark blonde wig forth as a symbolic replacement. DIAMONDS & CHANDELIERS
In the mid 1990s, Sia Furler started her music career in the small Adelaide acid jazz scene, and she released her first, unsuccessful, solo album, OnlySee in 1997. A move to London in the early noughties led to her appearance as a featured singer with the UK trip-hop band Zero 7, and the release of a second solo album, Healing is Difficult, which spawned a top 10 UK single, and an APRA Breakthrough Songwriter Award win in 2002. Sia’s third album, Colour the Small One, was produced by Jimmy Hogarth (Suzanne Vega, Duffy) and features a song co-written with Beck Hansen. Sia relocated to the US in the mid-2000s, and her 2008 album, Some People Have Real Problems, again produced by Hogarth, featured a song written by her and Kurstin, Death By Chocolate. This latter collaboration, and the marginally elevated chart positions of the album, were signs of great things to come. The birth of Sia, the mega-star, came with the release of her Kurstin-produced fifth album, We Are Born (2010), which did commercially far better than any of its predecessors, reaching No. 2 in Australia and No. 37 in the US, and was honoured with ARIA Music Awards for Best Pop Album and Best Independent Release. In 2011, two megahits she co-wrote and appeared on as a featured performer turned Sia into a household name: David Guetta’s Titanium, and Flo Rida’s Wild Ones. Sia also co-wrote Rihanna’s smash hit Diamonds, and
in the last few years has had her ‘writing machine’ hands in several other worldwide hit singles by the likes of Christina Aguilera, Ne-Yo, Rita Ora, Celine Dion, Britney Spears, Pitbull and Jennifer Lopez. Sia even found time to act as the executive producer of Kylie Minogue’s album Kiss Me Once, released in March 2014. Continuing this upward curve, Sia’s sixth solo album, 1000 Forms of Fear — referring to her own intimate acquaintance with fear — has enjoyed stratospheric success. Released in July of last year, it has been No. 1 in many countries around the world, including Australia and the US, and given rise to four hit singles, chief among them Chandelier, which has become one of the bestselling songs of 2014. A “whirlwind” is a fitting way of describing not only Sia’s presence in Kurstin’s studio, but also her chart performance during the first half of this decade. A HIT WITH THE LADIES
Greg Kurstin has a reputation as a star-maker, particularly with female vocalists, having masterminded the international breakthrough of Lily Allen, and co-writing and producing songs for the likes of Kelly Clarkson, Pink, Beyoncé, Ke$ha, Ellie Goulding, Lana del Rey, Lykke Li, and Katy Perry, as well as Santigold, Foster the People and Billy Idol. “I do also work with men!” Kurstin joked. “It just so happens that when you’ve had success with one artist, in my case Lily Allen, you get approached by record companies who say, ‘we have this female artist here,’ and want you do something similar again.” For the 57th Grammy Awards, Kurstin received his second Producer of the Year award nomination (his first came in 2010), for hit songs and albums with seven of the above-mentioned female singers. Slightly idiosyncratically perhaps, given his current state as one of the world’s prime purveyors of pop, Kurstin has his roots in jazz music, playing piano and studying at the New School for Jazz and Contemporary Music in New York and the California School of Arts in LA. As part of the band Geggy Tah, Kurstin was signed in 1994 to David Byrne’s Luaka Bop label, and in the early 2000s he worked as a session and touring musician with the likes the Red Hot Chili Peppers, Flaming Lips, Beck, and others. Apparently it was Kurstin and Sia’s joint connection with Beck that led to one thing, and then another, as the producer explained: “I first met Sia in 2003, when I was on tour with Beck, because she was friends with him. We started collaborating and writing songs together, but nothing ever happened with them. These songs just sat on my hard drive, until she recorded Death By Chocolate, which was the first public release of something we did together.” A STAR IS BORN
Two years later, Kurstin produced Sia’s We Are Born, which was described as “a massive pop leap: a sunshiny, highly caffeinated set of frothy dance tracks and feel-good lite-funk.” The producer and artist co-wrote five of the album’s 14 tracks — four
others were co-written by Sia with Samuel Dixon, a UK producer and songwriter and Sia’s live bassist. Four years on, Sia and Kurstin worked together again on 1000 Forms of Fear, co-writing six of the album’s 12 tracks. Kurstin also produced nine tracks, co-produced the three others, played 95 percent of the instruments, engineered most of the album, and even mixed seven of the songs. We’ll come to Kurstin’s singular tour-de-force approach to production further on. First, the American traced the gestation process of the songs. Kurstin: “Most of the time what Sia and I do is very spontaneous. She or I don’t really come in with a plan. I’ll usually sit at the piano, sometimes with a bass, and I’ll play a chord progression, and she’ll either ask me to change something, or will start singing a melody straight away. As I said, we both like to be decisive, so once we have a chord sequence and a melody we stick to it. We don’t really fuss over it too much. After that, we’ll both go to our computers. I’ll start building the song arrangement, playing or programming drums and then the rest of the arrangement and creating a vibe, working with headphones so as not to disturb her. Sia will be at her laptop, thinking of what the song is about and writing the lyrics. A lot of the time she will have recorded our initial attempts on her phone or computer, into a program like Garage Band or something, and she’ll listen to what we did and work with that.
Playing everything myself, it’s just faster for me. I know what I want to hear, and I know I can play what I hear
”
“Sia mostly does the melodies and the lyrics, and I’ll stick to the chords, the arrangements, and the melodic parts the instruments like guitars and keyboards might play. IT’S ALMOST LIKE A RACE TO SEE WHO FINISHES FIRST! SHE’LL WRITE THE LYRIC REALLY QUICKLY, SHE COULD FINISH A SONG IN 10 MINUTES SOMETIMES, AND I WANT TO BE READY WHILE SHE’S ON FIRE. At other times, if we have settled on a chord
progression, I may record that and loop it, and then play her the verse or chorus over and over again so she can try things out. Once we’ve written the song we’ll record it in a few hours, and sometimes that will end up as the finished version, maybe with a couple of fixes and additions later on. “Burn The Pages is an example of a song we did that way. I think I was playing the right handpattern and the bass with my left-hand, and she came up with the melody very quickly and then immediately wrote the lyrics. The original version sounded pretty good, and we tried recording that AT 17
song again ‘properly’ for the record, and even had Manny [Marroquin] mix it. Then later we went back and tried to match the demo, and ended up using a lot of the original demo parts. Cellophane came out of me using a delay plug-in on the bass to create some kind of pulsating, repetitive rhythm, and then I quickly jumped to the piano. That was one of the first songs we wrote for the record. When we wrote Fair Game, we didn’t think much of it. I contributed a little bit to the melody, but we threw it out after a day. But a week later she said, ‘you know that song we did the other day, let’s finish it.’ We thought that initial day was a waste, but it turned out to be one of our favourite songs. “There are a few songs on the album that were written by Sia, myself and a third writer, and they all came together in a similar fashion: starting with a demo recording made by Sia and the third writer. Free The Animal started with a really cool synthbased track made by Jasper Leak, to which Sia and I wrote an additional section. I changed it into a little bit more of a guitar-based track, with the synth arpeggios played by a guitar. Fire Meets Gasoline was done by Sam [Dixon] and we changed some of the chords. It was similar with Dressed in Black — we added some new sections to Grant Michaels’ demo. We didn’t write additional sections for Chandelier, which was written by Sia and Jesse Shatkin. Jesse used to be my assistant engineer, and his track was really great so I wanted to keep all the elements, which consisted mainly of programmed drums and synths — it really had a vibe. I added some organic elements to the arrangement, like guitar and piano and real Mellotron, to make it fit better with the sound of the rest of the album. We also redid the AT 18
vocals, because I thought it could be a single, and I wanted the vocals to be deeper and more present.” ONE MAN BAND
As intimated above, Greg Kurstin is the consummate DIY writer and producer. On 1000 Forms of Fear he played almost all the instruments on all 12 tracks, the only exceptions being Shatkin’s drums and keyboards on Chandelier, the main hook guitar on Hostage, which was played by The Strokes’ Nick Valensi, and the drums and some of the keyboards on the second biggest single from the album, Elastic Heart, which were programmed by co-producer Diplo. Kurstin’s instrumental credits include drums, bass, guitar, Mellotron, piano, xylophone, Chamberlin, as well as keyboards on a few tracks. The sound image of 1000 Forms of Fear is predominantly “organic”, as Kurstin calls it — ie. featuring mostly real instruments. With Kurstin producing, engineering and mixing, his own Echo Studio really is a one-stop facility. While he occasionally travels to other places, he far prefers to work at his own facility, where he knows the equipment and instruments inside out. And having started out as a musician, Kurstin still considers himself an engineering dilettante, which is another reason he’s more comfortable working in his own environment. Kurstin: “For a long time I didn’t really know what I was doing with respect to engineering, and as a result I did everything in-the-box in Logic. Over the years I have progressed a lot and my engineering skills are much better now. I’m still a Logic guy, and I use ProX, which I really like — I use the Track Stacks function a lot, and the general
organisational aspect of it is really great. I don’t have a desk, everything goes into Logic via my two Apogee Symphony I/Os. I also have a lot of hardware now — mic pres, outboard, and speakers, like my Adam S3X-Vs. “So I’m no longer 100 percent in Logic. I also have a drum kit here, bass and guitar amps, a piano, a Mellotron M400, a Chamberlin, and a whole bunch of hardware synths, including Moogs, ARPs, an Oberheim Matrix 12, Sequential Circuits Prophet 5, Vako Orcherstron, Roland Jupiter 4 and Juno 60. I could go on talking about them for days! But I didn’t use any of the outboard synths on the Sia record other than the Juno 60 on a few songs. MANY OF MY FAVOURITE IN-THE-BOX SOUND SOURCES ARE SYNTHS LIKE U-HE ZEBRA 2, LENNAR SYLENTH, SPECTRASONICS TRILIAN, AND VENGEANCE SOUND VPS PHALANX. And Logic’s EXS24 is my main sampler,
which I use a lot for programmed drums. Kurstin emphasised that all his instruments and microphones are “always plugged in and ready to record,” but when asked for details of his recording setup, his non-engineering background showed in the fact he couldn’t remember the names or model numbers of many of the pieces of kit in his studio. He ended up walking around his studio with the phone and a flashlight in his hands, explaining, “my drum mics consist of two Coles 4038s for overheads, Shure SM57s on the snare top and bottom, AKG D12 on the kick, and during the recordings of Sia’s record I had Sennheiser MD421s on the toms, but now I have AKG C414s — the old, silver ones. “My mic pres are… let me walk to my rack and shine a light so I can have a look… I have API
Kurstin processed the main bell synth you hear at the intro with Soundtoys’ Decapitator to give it some harmonic weight, and Valhalla Vintage Verb to give it some space.
Chandelier was mixed by Manny Marroquin, but once Kurstin recorded Sia’s vocals through his Telefunken ELA M 251 into a Chandler TG2, through a Fairchild compressor and into the Apogee Symphony, they only needed minimal processing through a combination of Logic channel EQ, Waves’ CLA-2A compressor and the Lexicon Hall reverb. This was the same vocal chain that he used on his mix of Burn the Pages.
500-series mic pres, and two Neve 1073LBs and my two overhead and two kick mics go through my four Helios Type 69s. The overheads then go through a Manley Vari Mu limiter/compressor, and I have another Altec 1567a mixer which I send my two snare mics through before they go through a Urei 1176. I also use a distant Beyerdynamic ribbon mic on the drums, which goes through a Standard Audio Level-Or, and an important part of the drum sound on Sia’s record came from an RCA 77 ribbon, placed just in front of the kick. On some of the songs it was the only drum mic I used! “I played my red Hagstrom bass on the entire album — in fact, I use it on 99% of the records I do. Sometimes it’s just DI’d, at other times I use my Ampeg B15S amp, on which I have this weird CAD E200 microphone. It’s similar to a Neumann FET47, but very inexpensive and it sounds good. It goes through a Helios Type 69 and then into an 1176. For electric guitar I used my 1960s Fender Jazzmaster a lot, and sometimes my Vox Phantom, and occasionally my 1966 Fender Telecaster. My main guitar amp is a Fender Deluxe Reverb, and I
record that with an SM57 and a Royer ribbon mic, going into my Altec 1567a, which submixes the two mics to one channel and then sends that through a Fairchild compressor. I record my piano with two AKG C414 mics, which go into two Chandler TG2 mic pres, and then into an Inward Connections Vac-Rac TSL4 tube limiter. I recorded Sia’s vocals with a Telefunken ELA M 251 — not a vintage one, but I’ve had it for a number of years — and that went through the TG2, into the Fairchild, and then the Apogee Symphony.” PLAY THE NUMBERS
While Kurstin’s one-man band act is extremely impressive, it does also raise some questions, like why he doesn’t occasionally get some other musicians in, just to mix things up? And also, how he manages to do so much of the engineering himself as well as playing. He laughed, “Well, because it’s easier for me this way! But I do work with two assistant engineers, Alex Pasco and Julian Burg, who help me out when necessary. When overdubbing I used to put things on loop record
and then run into the recording room, but now Alex often helps me record my own parts. Also, I’m not keen on doing stuff like crossfading and other detailed work in Logic. I’d be very happy if I never have to do those things again! So my assistants do them for me. “But with regards to me playing everything myself, it’s just faster for me. I know what I want to hear, and I know I can play what I hear. I hear a guitar part in my head, and I can just go and do it, so why get in another player? Plus I enjoy it. I’m a musician, and the thrill for me is to play. I suppose that’s the bottom line. Of course, I also enjoy playing with other musicians, and I love producing like that as well. I do that sometimes, getting in a bunch of musicians to record a track. But it’s just lots of fun for me to do it all myself. I GREW UP WITH FOUR-TRACK RECORDERS AND BUILDING UP ARRANGEMENTS ALL BY MYSELF. IT GIVES YOU THAT INSTANT GRATIFICATION.
“In addition, a lot of the time it’s just me and the artist in the room, and many artists like that. It means we can really concentrate and make quick AT 19
Kurstin’s initial mix of Burn the Pages became the final mix. The self-declared engineering dilettante mixed entirely in-thebox in Logic Pro X. Purple tracks 26-30 show a live drum loop played in by Kurstin. He chopped the beat up because the heavy compression was making the hi-hat pop out. Tracks 67-68 show a white sweep noise Kurstin injected into the chorus, and the plug-in chain here is the patch for the pluck sound that ends the song (track 8).
progress. Plus, Sia really likes my drumming. I’m sometimes a bit hard on myself when I feel I can’t quite play what I have in my head, but she likes my basic approach; just pounding the kick and the snare, and sometimes not even playing the hi-hat very much.” Kurstin’s talents also extend to top-level mixing. While he often uses the services of star mixers like Manny Marroquin and Serban Ghenea, he also mixes many of the songs he produces, including a fair number of hit singles. In the case of 1000 Forms of Fear, however, all the album’s singles so far released, including Chandelier and Elastic Heart, were mixed by Marroquin. Kurstin’s Logic screen shots of Chandelier show 70-odd tracks, which are a mixture of tracks from Shatkin’s original demo and Kurstin’s overdubs. “Jesse’s stuff is everywhere,” explained Kurstin, “like the ‘snap’ and ‘trap’ and ‘tap’ drum tracks at the top, and below that the ‘basses print’ and ‘perc’ and ‘bells’ tracks are his as well. “All the ‘pizz’ and ‘string’ tracks are also Jesse’s, probably from the ESX24 sampler — he uses Logic too — and I sent them out again to a guitar amp to add a bit of grit and make them sound dirtier. The pink tracks in the middle are me playing low notes on a Les Paul electric guitar. There are quite AT 20
a few muted tracks, which would have come from Jesse’s session. I replayed them on piano and other instruments to get the song to sound in line with the rest of the album. The vocals are at the bottom, below the ‘pre wavy synth’ track — they’re all the tracks with numbers in them. Just outside of the screen shot is the big reverb snare track you hear when she sings ‘1-2-3’, which has nine plug-ins for my rough mix, including a Lexicon reverb plate plug-in, another plate and several EQs. Manny then did a mix which added another dimension. I love that he keeps the energy and vibe of my demo mixes, and then manages to make them sound wider, and punchier and more 3D.” BURN THE PAST
Sometimes mixes go through a cycle of variations before coming full circle. In the case of Burn The Pages, the song had a number of incarnations, one of them mixed by Marroquin, before the decision was made to go back to an earlier version. “When I mix, whether it’s a rough or a final mix,” said Kurstin. “I work entirely in the box, using a mixture of Logic and non-Logic plug-ins. I do the best I can, but I don’t really think of my mixes as finished mixes. Though sometimes a mix falls into place, and we have a case of demo-itis and it becomes
everyone’s preferred version. That happened with Burn The Pages. The managers really loved the first mix Sia and I had done. So we went back to it in the end, adding a few things, but overall trying to change as little as possible. “The first version of Burn The Pages had a synth bass sound Sia didn’t like, because it sounded too electronic, so I tried to recreate it with a piano to give the low end more weight. There’s still a Juno 60 bass in the track. The five purple tracks, No. 26-30, are a live drum loop played by me. I compressed it strongly and the hi-hat sometimes popped out, so I chopped the beat up, instead of automating the hihat. There’s also a white noise sweep in the chorus, on Tracks 67-68. A bit below that are the vocals, ‘Old Vox,’ which are cleaned up overdubs. There were some timing issues with ‘t’ and ‘s’ sounds, so we locked them in a bit more. In my final mix, the lead vocals had a channel EQ, Waves CLA-2A, and a Lexicon Hall on it.” Sometimes dredging up the past can be difficult, but like a song that’s been hanging around too long, the release can be cathartic. By ‘burning the pages’ of many aspects of her past on 1000 Forms of Fear, Sia looks set to whip through the years to come like a whirlwind.
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FEATURE
Art meets rock at Sugar Mountain. Except for headliner Nas, where hip hop meets hipsters. Confused? Get in line. Story: Christopher Holder Artist Photos: Andrew Bibby
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Demographers keen to determine the exact size of Melbourne’s hipster population need only have taken their clickers to the turnstiles of January’s Sugar Mountain music festival. Brunswick Street’s cafes remained eerily quiet as hipsters of all genders and persuasions dutifully filed into the annual gathering where Art meets Music. Traditionally held in and around Melbourne’s Forum theatre, Sugar Mountain attempted to scale new heights, camping in the Victorian College of the Arts precinct, basking in the performing arts glow and revelling in the alleyways of the urban landscape. This is the hipster’s natural habitat. As is the Sugar Mountain way, the roster was eclectic, but heavy on the experimental, the ironic, the camp, the geeky… And then there was Nas. Hip hop royalty had come to town to play out his debut album, Illmatic. It was an unusual fit for Nas. But he headlined the main stage with one hour of intense, right-outta-the-projects sincerity. His performance was near flawless, but more like a straight-faced punchline to a day-long in-joke.
SWANS: GRACE UNDER PRESSURE Swans isn’t a band, it’s a music project; New York purveyors of experimental rock since 1982. Headed by Michael Gira on guitar, the current incarnation of the band features Christoph Hahn on lap steel, Thor Harris on percussion, Chris Pravdica on bass guitar, Norman Westberg on guitar, and Phil Puleo on drums. A Swans gig has always been an event. Gira traditionally likes his gigs loud, hot ’n’ sweaty. Just don’t head bang — Gira hates head banging. Back in ‘the day’ he was known to leap into the crowd and settle scores with head bangers. Lighting a Gauloise and perhaps curling your lip in time to the slide guitar drone might be more appropriate. Engineer Brandon Eggleston has worked with the band since its reformation in 2010 to perfect the best guitar and bass mic/cab configurations. It’s a settled backline setup now, although he didn’t bring all of his favourite pieces with him to Sugar
Mountain, with some mics replaced by, you guessed it, Shure SM57s. Here’s Brandon’s ideal stage kit: Norman (Rhythm Guitar): Electrovoice RE16 on the top, and a Granelli 5790 ‘bent 57’ on the lower cab. “I get a blend I like; wiggle the faders until it’s right.” Christoph (Slide): A Beyer M88 on his 2 x 15, an Electrovoice RE20 and a passive DI. “He runs two separate amps: one doing a loop or drone, the second amp doing the melody. Mix those equal.” Michael (Lead Guitar, Vocals): Beyerdynamic 201 and M88. “I rely more on the 201. The Orange Thunderverb 200 head provides the dirty meat ’n’ potato sound. A bass amp (GK1001) goes through a Mesa Boogie cab to provide clarity in the low end.” Chris (Bass): Sennheiser e906 on an Ampeg bass cab. It’s a huge guitar sound, in fact it’s 115dB of backline at the edge of the stage.
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NO ZU: NO PROBLEMO No Zu is a high-energy 11-piece band out of Melbourne, with loads of percussion, horns, vocals along with bass and guitar. It’s party music delivered with rave party intensity. Engineer Nao Anzai enthusiastically takes on the challenge. “I don’t bring a show file, just a channel list. The first song is all about fine tuning gain. The percussion is almost loud enough from the stage so I’ll work the vocals, the bass and drums, and after pulling a mix I’ll see if we need anything much from the percussion mics. The band is about raw energy and the audience expects a dub mix. It’s hard to pull a dub mix on a digital desk, I find. I’ll create a big reverb and a big delay, send vocals and horns to that, mainly, and create a return for those. I can ride the return to help create a dubby mix.” By the end of the set, Nao had reined in the mix — from seat-of-the-pants scary to controlled mayhem — without sacrificing any of the energy. Nao’s currently recording No Zu’s second album. The band made a limited edition cassette release of their previous album. I’m liking this band more and more.
RRR-EMOTE MIXING
Melbourne public radio station, RRR, was on site to record and do a six-hour live broadcast. They’d recently invested in a Midas Pro2 console and had set up behind the main stage. Taking a split from stage Dan Moore and Lachlan Wooden took care of the mix for broadcast, while also making a multitrack recording to Pro Tools along with a two-track master. The mix was sent via copper to the on-site RRR broadcast position with a microwave backup. This is all as it should be, but making lives tough was the stage spill. The open-air RRR mix position was so close to stage that it was very difficult to get anything like good isolation — even mixing on headphones was unreliable. So, using the Midas iPad remote mixing app, the guys set up a robust wi-fi router, and planned to head to a car with a decent stereo, tune into 102.7MHz on the FM dial, and cross-check the mix with what was going live to air. From there, they could fine tune the stem mix from the sweetspot comfort of the driver’s position. Genius! AT 24
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HIGH RISE WHINGERS
There are some monolithic figures in Australia’s music industry. Giants such as Michael Gudinksy, Michael Chugg… and The Melburnian. Çhe? The residents of The Melburnian apartment block in St Kilda Rd are renowned for putting a dB dampener on any gigs at the Sidney Myer Music Bowl. Bunch of chardy-sipping whingers. Sugar Mountain was also staged under the disapproving shadow of The Melburnian and with that came some stringent noise restriction handcuffs – 94dB at FOH, in fact. JPJ Audio took care of the PA requirements. Coincidentally JPJ was also operating over the road enriching the lives of The Melburnians by providing sound for a Kooks gig at The Bowl. JPJ’s Bass Gauci oversaw the Sugar Mountain spec, providing a d&b J rig for the main stage. The PA shoots down South Melbourne’s Dodds St, so the site is narrow with plenty of brick walls to avoid, which JPJ did masterfully. Single delay stacks of J were placed down the street, which came into their own when Nas drew all attendees to the main stage. Ground-stacked subs included pairs of Infrasubs along with the J-Subs. The LF spill was remarkably minimal at the back of the stage. I’m sure the good burghers of The Melburnian will be sending Bass a congratulatory note and a box of Quality Street. Stage 2 gave Bruce Johnston’s beloved Nexo Alpha rig a chance to stretch its legs. The main stage FOH package included Avid Profile mixing and Dolby Lake Processors. On-stage floor monitoring was an all-d&b M series affair. AT 26
Clockwise from top left: Keeping nicely within the limits for the sake of serial complainers at The Melburnian; Nas headlines the show with a throwback performance; and How To Dress Well’s Tom Krell provides a tip: Need a standard vocal sound and a sound with a big reverb? But you don’t have your regular FOH engineer with you who knows your songs? Set up two mics: one with, one without.
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FEATURE
Audio is so integral to Alien: Isolation’s suspenseful gameplay, it demands the player keeps their ear cocked for survival. Story: John Broomhall
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Extreme unease, spine-chilling suspense, oppressive fear, sudden terror — all in a day’s work for Byron Bullock, James Magee, Haydn Payne and the rest of their audio team who, over the last two years, have had the gratifying vocation of peddling audio anxiety for a living. The critically lauded, yet terror-inducing fruit of their labour is Creative Assembly’s horror title Alien: Isolation. It’s a videogame incarnation of a world set in a time just after Ridley Scott’s Alien. The movie was famed for harnessing the power of sound to frighten the living daylights out of moviegoers, but hearing an alien walk the same Sevastapol space station corridors knowing it’s hunting you, that invokes a new level of fear. It’s so scary, players recording YouTube walkthroughs of Alien: Isolation, who know they’re not actually in the game themselves, spend hours in a virtual crouch position peering out of doorways and over in-game furniture listening for the sound of their tormentor. KILLER SOUND
Byron Bullock: I’ve never played another game that encourages you to think about audio so extensively. The alien uses sound as a primary hunting sense, so you always need to be thinking about how much noise you’re making. We even track your real-world sound using the Microsoft Kinect and Playstation camera hardware, so if you scream on your sofa you’ll give away your position in-game! You’re constantly questioning if it’s safe to run. Are you going to attract the alien if you use a weapon? Plus sound can work to your advantage as a distraction — say deploying an IED or just hitting things with your claw axe. Imagine your way is blocked by a group of hostile humans. Make some noise, then hide whilst the alien comes and rips that group apart for you! Of course, now you have to safely evade the alien yourself. We encourage players to listen carefully; sound is one of your greatest weapons and best lines of defence. It can tell you how far away the alien is, if it’s in the vents or on the same level as you. Its vocals can communicate its awareness and the motion tracker reveals any movement around you with a familiar and disconcerting blip. AT: Did you define an overall ‘audio style guide’? Bullock: The art direction was tagged LoFi SciFi, a term used to describe the analogue, push-button world of Alien. The game is set just after the events of Ridley Scott’s 1979 movie, one of the first to portray space in this used-up, industrial ‘truckers in space’ way. It’s a time of CRT monitors; no holograms or flat screens. This obviously dictated the audio direction, everything needed to have that tactile feel and lo-fi vibe. Some of the most iconic sounds in the film are things like the mechanical ticks and beeps of the ship’s AI. AT: How does the audio in a game like Alien: Isolation actually operate? Bullock: We can play around 100 voices at any one time. A voice is one sound, a footstep for example, a door close or a dialogue line. THIS MAY SOUND LIKE A LOT BUT A GUNSHOT MIGHT BE MADE UP OF MULTIPLE VOICES; THE MAIN SHOT SOUND, MECHANICS LIKE THE
Wwise CAGE Match – The team built sound networks into the game so the alien’s artificial intelligence ‘hears’ sound in exactly the same way the player does. Make a loud move, and it’s coming to get you.
TRIGGER, A 5.1 REVERB AND REFLECTION TAIL, BULLET DROPS, LOW FREQUENCY SWEETENERS, PROJECTILE FLIGHT, IMPACT SOUNDS, THE CHARACTER’S CLOTHES, A VOCAL REACTION AND ANY OTHER EFFECTS LIKE EAR WHINE. It
adds up! We can place sound in the game-world in realtime, attach sounds to objects that move, and mix all those voices live at run-time using various realtime DSP like EQ, compression, limiters, etc. We can script logic to determine how and when sounds play and deliver mono through to 7.1. Game sound is very non-linear, highly interactive and dynamic compared with film. In film you know exactly what’s about to happen; you can set up moments, and sculpt the soundscape around them. That’s very hard to do in a game. Mostly you don’t know if and/or when an event will trigger; so you have to create systems, rules and logic that work dynamically. This is especially true in a game as interactive as Alien: Isolation. We aimed to sculpt audio the same way you would in a film to create the high tension and suspense so crucial to horror — but we had an unpredictable, cleverly adaptive alien AI using its senses to hunt the player down. It could hear the player, see the player and any light emitting from their torch. THIS WAS A PROBLEM FOR THE SOUND DEPARTMENT. WE NEEDED TO KNOW WHEN THE ALIEN WOULD ATTACK AND PREEMPT THAT MOMENT IN ORDER TO BUILD SUSPENSE AND TENSION.
WWISE CAGE MATCH
James Magee: We have several audio systems at work, designed to enable an immersive, reactive and dynamic sound experience, responsive to this emergent gameplay. We used Audiokinetic’s Wwise sound engine and developed bespoke tools to interface with it. Scripting was done through a powerful in-house game editor developed for the project called CAGE. This gave our sound designers the same tools as the level designers, enabling complex visual scripting and access to real-time data to drive the sound. One of our most significant audio systems was
based around the concept of sound networks, a way of intelligently and automatically subdividing the world into different acoustic spaces. Sound networks were primarily a means of ‘cheaply’ calculating high-quality Obstruction/ Occlusion, but also allowed us to attach parameters to run other environmental systems such as reverb and ambience. Our AI systems also made use of sound networks to affect logical sound representations, meaning they ‘heard’ sound in the same way as the player. We made use of this through Stealth and Threat parameters calculated from the AI’s awareness level. For example, as the player makes noises the alien can hear it based upon the surrounding environment. This real-time data was incredibly useful in driving many RTPCs (real-time parameter controls) within Wwise to emphasise mix elements, eg. bringing the sounds of the main character Amanda and the alien to the fore while ducking or filtering ambient sound to create focus and anticipation in certain moments. MUSICAL LAYERS
AT: How do you integrate music soundtrack in a world where environmental sounds are so important to the gameplay? Bullock: We constantly assessed how each sound would make the player feel. That often meant sounds would take on tonal and sometimes musical characteristics. Early on, we made a conscious decision not to have wall-to-wall music. That’s not to say we didn’t like the music — our composers did a fantastic job! — but we wanted the soundtrack to have contrast and dynamics. You don’t get used to music playing, so it has impact when it does. Some of the best audio moments are quiet ones without music — you walk into a dark corridor, it’s all quiet except for something moving in the dark. All you can hear is yourself, the walls creaking and some unexplained movement, it’s really tense. Also, sound can add a sense of realism which can
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SHOCK PEOPLE. THE SOUND OF THE ALIEN RIPPING APART SOMEONE AS THEY SCREAM LOUDLY FROM THE OTHER END OF A DARK CORRIDOR WILL INSTANTLY PUT YOU ON EDGE. THAT MOMENT DOESN’T NEED MUSIC — IT WOULD FEEL LESS REAL.
Magee: Using parameters such as Stealth and Threat, plus ‘Obstructed_Distance_To_Alien’ gave us tools for a highly dynamic music system. We wrote a custom Wwise plug-in that allowed us to pack three stereo tracks into a six-channel file, which can be mixed at runtime. These tracks were composed ‘vertically’, so when a threat draws closer or becomes aware of the player, the music layering system can dial up the suspense. Haydn Payne: During encounters with the alien we usually play one of our pieces of suspense music — mostly orchestral but with some synthesised/ sound design elements. We fade each of three group layers up and down in real-time according to game events. For instance, if the player is hiding in a locker, the music will sound almost ambient with the alien in the distance, but as it approaches, the intensity of the music rises and the high strings will become loudest as the alien walks right past you. If the music always responded directly to the alien’s proximity it could harm the game-play by giving away his position too easily, so throughout the game we change how we control the parameter that fades the levels up and down. One of the alternatives is the music reflecting the degree to which the player is being stealthy by using code to calculate how close unaware hostile enemies are, increasing the tension level as the player sneaks close to them. We can also script the music to change in different ways when the player sees particular objects or characters — say playing a sting when the alien walks round a corner into view, and subtly increasing the volume of the music as the player approaches a particular part of the environment. A lot of this is controlled by the enemies’ AI state. As the alien charges to attack we usually trigger a short music riser which builds up to the point when the alien reaches the player. We also use the AI state of the hostile humans to control action music, which has three levels of intensity we switch between based on how aware the hostiles are and how much danger the player is in. Bullock: We’re constantly asking ourselves, what should the player feel and how can we achieve that? For instance, layering subliminal non-literal sounds underneath real-world sounds to give them an emotional connection; like adding alien hisses into steam blasts and doors closing, or alien screeches into fire bursts and metal corridor groans. We did that sort of thing a lot. It helps to keep the player on edge. The player thinks they heard the alien — or did they? Sound can be a great tool for misleading the player and at the same time it’s great for doing the opposite, leading the player. Sound can direct the player to a certain area or tell you what’s happening behind a door, like someone screaming in the next room. All these off-screen sounds can tell a story in their own little way and ultimately contribute to the greater narrative. AT 30
“
The Wwise layer splitter that ramps up the music relative to how dead you’re about to be.
We even track your realworld sound, so if you scream on your sofa you’ll give away your position in-game!
”
The creepy sound design team that gets a kick out of scaring you half to death. Left to right: Jack Melham, James Magee, Byron Bullock, Sam Cooper, Stuart Sowerby and Haydn Payne.
RAIDERS OF THE LOST ARCHIVE Bullock: We tried our hardest to get as much original material from the film as possible but you have to remember Alien was made a long time ago on analogue reel-to-reel machines. Fox was very helpful and sent someone down to their archives deep beneath the LA Fox lot to dig up a box marked Alien, which contained an 8-track analogue tape reel. No one knew what was on it. We arranged for it to be digitised and sent over electronically — super exciting!
Opening the tracks in Pro Tools, to our surprise and delight we found some of the original sound effects used to underpin the film score. It was a gold mine! One of my favourites was what we eventually dubbed the ‘Space Whale’ — a low moaning-type sound you hear over Alien’s opening logo. I love that we’re able to introduce this original content that’s sat there for 35-odd years to a new audience. It really gives the game an authentic feel.
KNOWING THE SCORE AT: What’s the division of labour amongst the composing team ? Alexis Smith: Christian Henson initially concentrated on the orchestral elements, while Joe Henson and I focused on the more electronic side. Although there was a lot of cross-fertilisation. We worked on it for about a year and a half and composed about 200 minutes of music in total, but most of it is in a kind of interactive kit form, as the game-play can be 7-20 hours, depending on the player. Joe Henson: We’re all massive fans of the original film; one of the few we still watch a couple of times a year. It was a great honour to be able to use some of Jerry Goldsmith’s themes, as it never really works when you pastiche something as familiar as Alien. Alexis Smith: Creative Assembly licensed about 10 minutes of the original score that contained the most recognisable themes, which we then re-arranged and re-recorded in various ways. AT: How did the interactive nature of the music score affect your approach? Joe Henson: We always start off by writing a piece that can stand up by itself. After that we can work
with the interactive engine. It is a bit of a puzzle, but one that is really rewarding when it works well! We recorded a lot of it in our own studios. We have a large collection of analogue synths and weird homemade instruments and try and play as much as we
can ourselves. The orchestral recordings were done with Jake Jackson at Air Lyndhurst with the Chamber Orchestra Of London. We were lucky enough to have some of the players from the original score, who were very informative on some of the techniques used.
MAKING MOVES: RECORDING FOLEY Alien: Isolation’s foley was recorded at Shepperton Studios by Head of Digital Recording & Editorial for the Pinewood Studios Group, Glen Gathard. You’ll have heard his work in movie blockbuster franchises like Batman, Bond and Harry Potter. Glen Gathard: It’s all about organically recreating the sounds needed to bring a scene or game to life, ‘performing’ the sounds as if you are the person onscreen. Foley can add to the emotion of the person you’re performing — Alien: Isolation is a perfect example. The footsteps across all surfaces were thought about from different emotional states — if you’re scared, the feet will be light, stealth-like with random imperfection — plus additive layers like scuffs and squeaks. The cloth and equipment layers work similarly — when hiding you want to keep noise to a minimum; having cloth and item detail heightens the silence — if all you can hear is your clothing, a little dog-tag rattle and light scuffs underfoot, everything feels quieter and it builds suspense — it’s more claustrophobic; and then the big scare moments with the alien feel much, much bigger. Shepperton’s foley theatre is one of the very best in the world. There’s nothing we can’t do, from high heels on wood, flooding the surfaces with water to dripping molten fire plastics from a step ladder. It’s a massive playground full of people that truly love the art of foley and sound design. I’m also a massive gamer lucky to cross the world of film and games quite frequently. Since working on games I’ve learnt so much, and hold videogame sound
designers in high regard. To create sound for a project that will be played over and over again — and to keep it feeling cool, fresh and dramatic is such a talent. FOLEY IMPLEMENTATION James Magee: Our foley system was an attempt to push the boundaries of what could be achieved with real-time information. The in-game stealth mechanics mean you’re aiming to make as little sound as possible while trying to survive, and dynamic foley really helped this feel credible. Real-time player input completely drives Amanda’s foley system through a combination of Movement_Speed and Acceleration RTPCs, rather than the traditional approach of tagging her animations with event metadata. For example, as the player moves Amanda around, we apply the Movement_Speed parameter to a blend container which crossfades between slow walking and running foley one-shot samples. Each time a footstep is triggered by code, we also trigger a foley sound event which activates the blend container, resulting in the right foley sound playing back. We detect the acceleration of the first-person camera movement so when the player quickly turns Amanda around to see some unseen horror behind her, the Acceleration RTPC triggers a foley sound through a Wwise switch to simulate clothing movement. It’s a subtle audio system but adds a lot to connect the player’s input to the character — especially effective for a first-person survival horror.
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TIPS & TRICKS: CAPTURING SOUNDS COMPUTER SEQUENCES Haydn Payne: Some of our computer sounds were made by creating a software synth that generatively plays sequences of beeps of random pitch, duration, waveform, and delay settings. We made long recordings of the synth’s output as it ran continuously and edited together any bits that happened to sound good. The resulting audio was too clean compared to the lo-fi analogue quality of the computer sounds from the film so we fed them through various tiny damaged speakers taken from broken toys and recorded the results. CREEPY FOOTSTEPS Payne: When the alien charges at you, the audio system detects its speed is above a certain threshold and triggers running samples for its movement. If the threshold is not reached it plays separate walk samples which are dynamically mixed quieter and with more low-pass filter relative to the movement EMA_AT102_[Press}.pdf 10/06/14 speed decreasing. Each walk footstep 1sound is made
from two parts — the claws hitting the floor and the weightier foot impact. When the walking speed is fast the two parts are played in quick succession but as the alien slows down, the delay between the two parts gradually increases — there’s a noticeable gap between them at the slowest movement speeds. ALIEN VOICES Payne: The vocal sound the alien makes when it has heard something suspicious is different from when it’s seen a target, or has lost track of its target and so on. The alien’s language in its different states can be learnt by the player to understand more about its behaviour. The sound design for this drew from a variety of source material including the sounds of big cats, reptiles, pigs, bears, birds, and the human voice. AUDIBLE ELECTRONICS Bullock: We used a coil pickup microphone which captures changes in electromagnetic signals and 11:40 AM turns them into audible information to record all
sorts of computer and static sounds. However, my favourite came from my fiancé’s heated rollers. She has a machine that heats the rollers instantly using induction technology. I put the microphone inside this contraption and got the most insane electrical blast sound which we used in several places, most noticeably in the reactor levels as part of Sevastapol’s energy core. DRY ICE INTERFACES Bullock: We used contact microphones and a hydrophone to capture dry ice and metal vibrations for Sevastapol’s metal interior and did some circuit bending to generate interesting electronic sounds for interfaces. CONDOMS & INTERNAL ORGANS Bullock: We submerged a trusty SM57 (inside a condom) into a bucket of slime (lots of different food and drink mixed together) and got some great internal organ sounds for the alien.
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REVIEW
AEA NUVO N22
Ribbon Microphone AEA’s new phantom-powered ribbon mic sees the company continue to move away from strictly retro designs, and closer to the source.
NEED TO KNOW
Review: Greg Walker
PRICE $1339 CONTACT Mixmasters: (08) 8278 8506 or sales@mixmasters.com.au
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PROS Great tones on a wide variety of sources Copes with high SPL & phantom power Good amount of high-end detail for a ribbon design Modest proximity effect allows singers to get in close
CONS Somewhat noisy in very quiet applications Shock mount clip a bit cheap
SUMMARY A well made, sweetly voiced active ribbon that is optimised for close miking applications. AEA has got manufacturing ribbon mics down pat and the svelte looking N22 is a power-punching new addition to their range.
Ribbon microphone technology has come a long way in the last few years, with new models targeted at specific applications. It’s no longer just ‘getting a ribbon’; there’s higher SPL-capable models for getting right up on guitar cabinets, all the way down to phantom-powered models with higher outputs for those distant-miked mouse squeaks. AEA has been at the forefront of this evolution with the introduction of phantom-powered models like the A440 and A840. While the company’s roots are firmly planted in the fertile compost of old RCA designs — which was still evident in their newer phantom-powered models — with it’s Nuvo range, AEA has completely dispensed with the old RCA-style yokes, bodies and grills. The N22 active ribbon mic is the first cab off the rank, and offers a contemporary looking silver body with a broad silver ‘belt’ and a funky black cloth, cutaway ribbon enclosure. Looking like a cross between an art deco ray gun and a high-end Italian coffee-making accessory, the AEA N22 definitely has presence on the end of a boom stand. The subtly embossed silver Nuvo logo on the front is a nice touch and complements the more traditional red AEA logo on top of the mic. The twin logos are a great metaphor for where a company like AEA finds itself in the second decade of the 21st century — the N22 is a thoroughly modern mic made by people who love vintage gear. NUVO RICHE
The N22 is aimed squarely at the home-studioowning recording musician. In a Tony Abbottlike act of hard sell, the manual even describes its ideal target market singer-songwriter as ‘hard-working’. Outside of references to James Brown this is perhaps the first time I have ever seen the words ‘musician’ and ‘hard-working’ together in one sentence. On a more practical level, the N22 has been optimised for close-up miking of sources such as vocals, electric guitar cabs, piano and drum overheads while retaining enough gain to cope with more ambient room positions, and lower quality audio interfaces. The mic is built by hand in Pasadena, USA and utilises JFETs and a custom-wound German transformer. The N22 ships in a nifty black plastic case with an adequate but somewhat uninspiring mini-shockmount and an extended three-year warranty. AEA has thoughtfully enclosed a small cloth microphone sock that will protect the mic when not in use — these are particularly handy for ribbon mics whose large magnets attract small particles of all sorts. Over time these particles can build up and degrade the ribbon element, so ‘socking up’ the mic after hours is a good long-term maintenance policy. FROWN TURNS TO SMILE
Firing up the N22 for the first time on an acoustic guitar I was pleased to hear full-bodied tone with
plenty of life in the upper frequencies. The supplied chart shows a characteristically flat frequency response up to around 5kHz where the mic smoothly dips down to -10dB at 16kHz. Down low there is a gradual roll-off from around 300Hz. As is the case with all other ribbon mics, this classic ‘frowny face’ frequency emphasis means that the mic must do all its best work in the midrange, and the quality therein will decide whether it becomes a much used and loved studio tool or an occasional flavour in a recordist’s wider arsenal. My acoustic guitar recordings revealed a sweet round midrange flavour while there was clarity in the higher frequencies and an absence of ‘mud’ down low where many cheaper ribbons start to lose definition. In a much sterner test of the N22’s mettle I put it up alongside a Zigma C-Lol-47 condenser and a Shure SM58 on a vocal session with Richard Morrison (The Night Party). Richard is one of the louder singers going around with a lot of heat in the 1-4kHz range, and while both the other mics started to colour in a less than pleasing way on the louder passages, the N22 stayed sweet and strong all the way. I ended up using mainly the N22 with a little of the SM58 going through an amp and we got the right amount of smoothness and grit to carry the track perfectly. On another session I used the N22 for close miking a Fender Tremolux amp running pretty hot and again the N22 didn’t break a sweat and gave a nice blend of direct signal and (much quieter) room reflections through the two poles of the mic. Again the tone was super useable and sat well in the track with a minimum of fuss. GOING WIDER
Having earned a big thumbs up in close miking and louder applications, it was time to try the AEA in some more ambient positions on a choir recording in a large church as well as on various ensemble studio vocals where it was used as a room mic. In the church at a distance of 20m from a group of six singers I was happy with the tone but a little disappointed at the high noise floor despite trying several different preamps. There’s no published signal-to-noise spec in the otherwise excellent and informative manual. In every other application the S/N ratio wasn’t a problem but I would not recommend the N22 if you are going to do a lot of deep room miking or tracking of very quiet sources. For that, AEA offers another new ribbon (the N8 model) which is optimised for distant miking situations. On a more positive note, the N22 did great things on strings and percussion recordings as well as delivering great room tone on drum and guitar recordings. Close up on strings it gives a lovely warm, thick and vivid picture of complex violin and cello tones while there’s no shortage of guts and weight on double bass. On percussion it dodges the most strident upper frequencies that often require taming EQ using condenser mics, while delivering more useful midrange information than most. Since I first flicked the phantom power switch on the N22’s preamp it has made its way onto pretty AT 35
much every recording I have done over the last two months — either as the main close source or as a secondary room capture where it has blended well with dynamic and condensers closer in. ANNIE GET YOUR 22
My verdict on the AEA N22 is a very positive one. The mic is solid under pressure and performs a lot of standard recording duties with aplomb. Being optimised for close miking means it can shine in poor or limiting acoustic situations where other more sensitive ribbons might reveal too much at greater distances from the source. It also means a more controlled proximity effect than most other ribbons on the market. It has a notably even polar response and takes EQ very well allowing the user to bring all that lovely smooth top end into focus. I’ve yet to hear a bad microphone from AEA and the N22 lives up to the company’s considerable reputation for quality and vibe. While condenser microphones may be the ultimate studio allrounders, quality active ribbons like the N22 come a surprisingly close second in terms of functionality and versatility. If you’re looking for a significant other in the ribbon microphone market, the N22 is definitely worth spending some time with.
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REVIEW
MACKIE DL32R Tablet Rack Mixer Mackie pioneered tablet mixers and — now the market has come of age — the DL32R rack version and Master Fader update shows the value in getting there first.
NEED TO KNOW
Review: Mark Davie
PRICE $3499 CONTACT CMI Music & Audio: (03) 9315 2244 or sales@cmi.com.au
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PROS 24-channel recording via USB Master Fader app coming of age Onyx+ preamps sound great Plenty of I/O & DSP onboard
CONS Can’t rearrange fader order on the fly Headphone output goes where the rack does
SUMMARY Mackie’s DL32R tablet-controlled rack mixer builds on the successes of the DL1608 with more I/O and USB multi-tracking. And v3 of the Master Fader app shows Mackie is still ahead of the curve. The only thing you have to worry about is where to put it.
Since my last outing with Mackie’s first-tomarket, iPad-controlled live mixer — the DL1608 — a lot has changed. For one, a product that could have easily turned into a gimmicky, flash-in-the-pan pockmark in history, has spawned an entirely new market of live mixer products. Just off the top of my head, there are now half a dozen manufacturers in this space, with more certain to come. It’s not just a manufacturer-driven market though. Users are eating up these iPad-controlled devices, and for good reason too. Just as computers spawned the home studio revolution and plug-ins gave musicians previously unattainable control over their mix, tablet-based mixers have done the same for gigging musicians. The best a covers band used to be able to hope for was a portable analogue mixer with perhaps a built-in stereo effects engine. With the mixer on stage next to them, there was no way of tuning the PA, or at least ringing out resonant frequencies in the room, no way of getting a decent FOH balance, and typically no dynamics control or graphic EQs. Tablet-based mixers fix all of those issues; roaming mix capability, loads of onboard processing, and with leftover DSP, they often do a whole lot more, like USB recording and playback. I’m not saying it’s the only application for the DL, far from it but it has put the power back into the hands of the gigging musician, and they’re loving it. A friend of mine plays in a covers band that owns a Mackie DL1608, and it’s been a complete game changer for them. Of course, there’s no substitute for having a dedicated sound engineer mixing a show — indeed, the majority of gigs require it — and most would still rather physical faders for the job. But Mackie has been working hard on its Master Fader app, adding more pro features with each update. When it released the DL1608, Mackie also built a user forum for feature requests, and a lot of those have been implemented. So as well as being first to market, Mackie is already up to version 3.0.1 of Master Fader, and it shows. FADE UP
When I reviewed the DL1608, there were a number of annoying omissions, and one of the most requested features was the ability to link two consecutive mono channels into a bona fide stereo channel. Mackie implemented that fix in version two. Other v2 fixes included aux send linking for stereo outputs (like in-ear monitors or external feeds), pre-DSP aux source for non-effected monitoring, aux mutes, mute groups, view groups, and a quick access panel that lets you access effects mutes and delay tap tempos no matter which view you’re in. Mackie also added a master output mute, which was doubtlessly needed. In the early version, there was only an EQ select button on each channel, which brought up the large EQ GUI. You then had to swipe to the dynamics section, and again to the effects. Now you can access the dynamics view directly from the gain reduction meters.
Mackie also added 350ms of alignment delay for delay stacks in v2, with an accompanying translation into metres and feet. It had an adjustment for temperature compensation too, but that’s all been nixed in v3. Perhaps delay stacks were stretching the imagination for a DL series use case. The delay facility has been moved to the monitor section in the latest version, for time aligning your headphones or monitor speakers to the FOH rig.
drums with a sub group fader for instance. In v3 you can create Groups A through F, which can include any input, return, effect, sub group or VCA. Better yet, you have the same amount of customisable fader banks for each of the 14 auxes and for the effects sends too, giving you even more flexibility over not only your own mix, but what faders you can set up in each band member’s iPhone MyFader app too.
THIRD TIME LUCKY
Version three is another big step forward for the app. There’s some universal graphics changes — the faders now look like faders, instead of pointers, and all the circular slider buttons have been replaced with square ones. But the biggest leap is in how you navigate around the fader banks, especially with the larger DL32R. The default fader view can be a bit unwieldy at times. With only eight faders on the display at any one time, it can take a bit of side-scrolling to get through all 32 input channels, 14 aux outputs, three effects faders, six sub groups, six VCAs, and six matrix outputs on the DL32R. So here’s a quick look at the new views:
THE BIG PICTURE — There’s now a global channel view, which shows you miniature colour-coded faders with accompanying meters for every input, output, aux, group, VCA, effect, matrix out and mute group, all in one surprisingly easy-to-read window. From there, you can select a particular channel and be taken right to its indepth view.
WIDER VIEW GROUP — The other major navigation features are all in a strip down the right hand side of the mixer window. In v2, Mackie introduced View Groups, which allowed you to customise your own groups of fader banks —
MASTER FADER BANKS — You also don’t have to waste a custom fader bank to group all your VCAs, for instance, because there’s a separate master fader section. You can flick between showing all master faders, singling out your LR master, or displaying group, effect, VCA, aux, or matrix masters on their own screen.
MUTE GROUPS — At the top of this navigation side bar is a new Mute Group section, which allows you to set up six, nameable mute groups. All of these custom banks and groups are really easy to assign from a single window, and you can also switch between viewing multiple groups of faders together or just one bank at a time. While the new custom fader banks definitely give the iPad workflow a much-needed speed boost, I still have one bug bear to do with faders: You can’t swap fader positions. This is a real annoyance for me. Say you have your show file laid out with drums, bass, guitars, and vocals, then find out there’s an extra guitarist coming to the gig, or the drummer has three toms, but you only set aside two faders, you can’t simply part the waters and inject a fader from a far off land into the middle. This inability to rearrange your fader positions extends to the View Groups too. Although you might still be able to group all the drums or guitarists, you can’t rearrange them into logical orders based on where they are on stage or the size of tom. You’d have to go back and meticulously AT 39
Master Fader screens clockwise from top left: Channel assignments at a glance; the vintage dynamics options; inside the routing matrix; and the two halves of your USB stream, playback and multi-channel recording.
rename or load presets for each channel to suit. I’m looking forward to v4 for that fix. RACK YOUR BRAIN
The DL32R has 32 microphone preamps on XLR, with the last eight being combo jacks to allow for instrument inputs. The preamps are Mackie’s updated Onyx+ units and they sound good without clipping hard at the top end. They have plenty of presence, take EQ well, and with 60dB of clean gain, can handle any live source I can think of. Each channel has individual switchable 48V phantom power, and has a phase flip too. There are no pads on the channels, which I didn’t have a problem with in practise. There is a Trim control below the gain setting, but this is purely for digitally adjusting recording output levels. You can’t hit the preamps hard and back off the gain with the trim control like you can on a Midas. There are 16 outputs, also via XLR, with the last pair being dedicated master L/R outputs. And aside from the monitor output and headphone output with its own level control, there’s also a stereo AES digital output. The whole thing weighs just over 8kg, and only takes up a 3U rack space. That said, it is quite deep, AT 40
so you’d want to ensure you buy a long enough case if you’re planning on it being portable. The biggest question is where to sit the thing. The obvious choice would be in your stage rack. It’s essentially a wireless digital multicore, and you need your I/O as close to the stage as possible to save breaking out copper multicores. But then your headphone and monitor outputs are in the rack too, not to mention the talkback mic input. The iPad is only a controller, so it doesn’t output any audio from its headphone jack, or feed any to the rack via its microphone. Placement didn’t feel like a big issue with the DL1608, because it was smaller and came in that desktop mixer form factor. But the DL32R’s uncanny resemblance to a digital console’s networked stage box pretty much seals its position upstage. It would be nice if Mackie could come up with a wireless breakout interface with a headphone amp/monitor output and talkback input that you could park at your FOH mix position. The other obvious worry is the move to full wireless control. Unlike the DL1608, you can’t simply dock your iPad into the DL32R and get a hard-wired connection. That said, Mackie seems to have nailed down the synchronisation process,
and even if I forced the app to close, or killed an operation, Master Fader found and synced to the DL32R in seconds. I would recommend using a wireless-N, 5GHz router though, and setting up a backup network. Think of it as a virtual multicore that could just disappear into the ether. LONG TRACK RECORD
The other placement conundrum is due to the multi-track recording feature, accessible from the two USB connectors on the rear of the DL32R. You can record up to 24 tracks (up to 24-bit/48k) direct to a self-powered hard drive over the USB A connector, or 32 channels to a computer over the USB B connector. Both will be able to record 32 channels come April, according to Mackie. The assignment matrix allows you to pick either pre- or post-DSP for each channel, and it doesn’t have to be direct inputs, it can be a mixture of groups, auxes, effects returns, whatever. The caveat is you can’t use both USB connectors simultaneously, so you can’t back up your computer recording with the directto-drive method; the USB A port always overrides the USB B port. Keeping it all in a stage rack is fine if you’ve just got a hard drive hooked up; your iPad is the only
The DL32R from behind; plenty of ventilation, a wireless card that can be swapped out for a Dante expansion option, and the two USB ports, where the Drive port overrides the Computer port.
reference as to what’s getting recorded anyway. But seeing a DAW draw waveforms in real-time can be a big confidence booster, so you’d want to be close to your computer screen for that mode. So where should I put the DL32R now? When recording straight to a hard drive, the file format is multi-channel WAV files, with a maximum size of 2GB. It will automatically begin a new file, without gaps, each time the 2GB file limit is reached. So, for 24 tracks, that works out to be roughly 10 minutes per file. The manual says not to record to thumb drives either, and it was pretty right about that. I tried, and while it still managed to look like it was stably recording on the iPad, there were glitches in the audio on playback. A hard drive worked fine. Master Fader also lets you set up two inputs for each channel. By default, Input A takes its source from mic preamps 1-32, and Input B from USB returns 1-32; a classic virtual soundcheck setup. You can, however, set them up as alternate stage patches, or to switch to backup mics. And while there are global switches, you can also flip only the amount of tracks that are in your multi-track playback source. So if you’ve got an eight-channel backing track you want to play back for one song, you can just flip that amount of channels to the USB inputs, and flip them back after the song is done. You can also play back stereo files from the hard drive, but they must be 48k WAVs to match the session; no MP3s, 44.1k files, etc. Mackie also bills the DL32R as a studio interface, and while it definitely sounds clean enough, you’d have to watch the fan noise. While totally acceptable in a live gig, it’s definitely a unit you’d have to isolate in a studio environment. VINTAGE V MODERN
One fun feature of Master Fader is the ability to swap between a ‘Modern’ take of the EQ, gate and compressor, and ‘Vintage’ versions. You can mix and match for each channel, and it’s not just a different face plate either. The controls vary (often
more limited on the Vintage units), and they work differently too. The Vintage compressor has a program dependent release, and when the attack is set to fast, it’s fast. The Modern compressor has a nice operating range with the ability to squash and pump sounds if really needed with an inf:1 setting, 20dB of makeup gain, 0.1-300ms attack time, and 50ms to a long 3s release time. Of course, it does subtle compression really well too, with a soft knee setting and ratios gently exponentially rising from 1:1 as you scroll through the fader (ie. there’s lots of finer detail in the lower ratio settings). The Modern EQ has four bands, with individually tailorable Q, as well as a high-pass filter. The Vintage equivalent only has three bands — low and high shelving, with a parametric mid. It also has a high-pass filter, and all the controls are stepped as opposed to sliders in the Modern equivalent. On the outputs, you can switch between fourband parametric or 31-band graphics, allowing you to either quickly sculpt overall groups, or finely tune a system. There are two reverb and one delay effect engines on the DL32R. You can choose between eight different classic reverb types, five different delays including a tape echo, and there are plenty of preset starting points, or you can store your own. MASTER IT
If you’re seriously contemplating a Mackie DL32R purchase, the best thing you could do would be to download Master Fader from the App Store and get familiar with it. Just a brief browse of the I/O routing page alone will give you a quick understanding of how truly flexible this device is, including multiple device hookup over Dante. In practise, the device works extremely well. Every iteration of Master Fader feels like the system makes a big leap, and in a few areas, bounds past other digital console form factors. There’s something to be said for being able to navigate an entire session smoothly from a touchscreen.
“
Just as computers gave musicians previously unattainable control over their mix, tablet-based mixers have done the same for gigging musicians
”
AT 41
REVIEW
EMERGENCE AS8 PRO S Monitor System
Joe Hayes has spent 20 years refining a studio monitor system that suits the way our ears work, and it doesn’t require a custom ear mould. Review: Mark Davie The 60W sub uses a Wavecor Balanced Driver with a paper cone. The key technology Wavecor has on its side is a longer linear magnetic field around the voice coil than standard. It means the drivers behave linearly till the end of their excursion, right where the voltage is changing directions. The Hypex amps, which also house the DSP, are Class D with two parts per million distortion. Although not hugely powerful on paper, I found no issues with level in a typical near- to mid-field setting.
NEED TO KNOW
Under the QRD awning of each satellite sits a two-inch, aluminium cone, full range driver that puts out 30W of power. They’re capable of stretching down to 150Hz, but to stop the satellite’s resonating, they’ve limited the response to 250Hz for now. A recent prototype with 70g of silicon injected into the housing might allow the extension to go lower.
PRICE $995 CONTACT Acoustic3D: (07) 3177 3377 or info@acoustic3d.com www.emergencea3d.com
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PROS Diffuse hologram effect gives extra depth Flexible voicing Relatively cheap for a different monitoring solution
CONS No feedback of settings No Mac DSP control yet
SUMMARY The Emergence AS8 Pro S monitoring system diffuses at the source, so you can treat the sound instead of the room. It’s designed to put you back in the original acoustic of your recording, not your listening environment, with DSP to help the image down low.
It’s mid-morning at AudioTechnology HQ, and Joe Hayes rocks up with his latest invention in tow. He surveys the office’s midcentury bay windows, hardwood floor, timber desks, adjoining bass-trap of a kitchen and says, “This’ll be perfect” as he starts to pull out his sub-sat studio monitor system. Hardly perfect; the office’s only redeeming acoustic treatment is a bookshelf of old magazines, and the desk clutter I like to call ‘random diffusion’. Any other studio monitor manufacturer would laugh, cry, or promptly do a 180 and get back in their hire car. It’s definitely not a serious monitoring environment, and to be clear, we don’t use it as one. But to Hayes, it was just another emulation of a home studio’s crappy acoustics, and he could hardly care less. See, Hayes isn’t worried about your room. His monitor placement guide doesn’t come with a preamble on room acoustics or diagrams pointing out angles of incidence. He’s got a whole different way of thinking about studio monitors and diffusion; a relationship he’s been patenting since the ’90s. The general gist is this: rather than adjust the acoustics of your space to try and flatten out its frequency response at your listening position, why not just get rid of the room’s acoustic altogether? SPATIAL VERSION ONE
We’ve reviewed Hayes’ inventions before. One of our mainstay reviewers, Brad Watts, still has an early pair of Spatial One nearfields. Hayes’ new iteration under the Acoustic3D company name is called the Emergence AS8 Pro S, which I’m not convinced is a better name than Spatial One. It’s a sub and satellite combo, with conspicuously Independence Day spacecraft-looking speakers. As you can probably tell, just by looking at it, that is indeed an angled quadratic residue diffusor leering over each satellite’s driver. But it’s not just any quadratic residue diffusor, it’s a correctly tuned one, according to Hayes. What he means is, Schroeder’s original concept treated sound as if it had a planar wavefront, but sound radiates from a source with a spherical wavefront, like a ripple in a 3D pond. By Hayes’ calculations, your back wall diffusor would have to be at least 6m away from a source to have the desired effect. So putting a QRD right at the speaker’s mouth without doing something to account for this curved wavefront would be ineffective. On this design, he’s compensated accordingly to diffuse across a quoted range of 1.4-20kHz. So why diffuse at the source? Well, it’s a good question and one that requires a re-think about how our ears work. RIDING THE WAVELET
I was re-introduced to the story of Daniel Kish not long ago, the blind man who can ride a bike by echolocating with clicking noises. Well, it’s really his gimmick to entice morning shows into booking him to talk about what he really cares about. Kish’s crusade is to stop institutions telling blind people they can’t see. That’s right. He says blind people can see, if only we’d change
our expectations surrounding blindness. And it turns out, he’s right. When blind people train themselves to echolocate, the part of the brain they retrain is the primary visual cortex. It lights up like a Christmas tree as they ping out a car, tree, recording console, whatever. They see, just with a different methodology. After all, what we interpret as eyesight or vision is really only our visual perception of what’s going on around us based on processing the information contained in visible light. Hayes sees acoustics in a similar way. As much as he hates statistics, he reckons acoustics is all about stats. “Perception is the probability of what’s going on,” he says. And with normal studio monitors we create a perception problem. First, there’s the recorded acoustic, which is a sound source plus a bunch of statistical properties, including the number of reflections per second that might influence the general tonal quality of the reverb. “When you put that into your listening room,” says Hayes, “which has a whole new bunch of statistics itself, you end up with two sets of stats that the brain just can’t really work out.” So, to solve this mash-up of acoustic statistics, we try to adjust the response of our listening environment rather than ‘fix it at the source’. Hmm, where have we heard that before? Hayes has spent the last 20 years trying to prove why flattening the listening environment is a bad idea. But back to this idea of perception and how we hear. When Hayes finished university, he says the hardest thing he had to do was unlearn much of his acoustics training. In 1984, Jean Morlet modified physicist Dennis Gabor’s work to come up with the Morlet wavelet and later on proposed it could be useful for analysing music. When Hayes came across Morlet’s work it was all pretty recent, but it helped inspire his invention’s first iteration in 1990. A Fourier transform, one of the more common acoustic measurements, receives signals in a window of time and gives you a picture of what happened in that window. So, you might have had 10kHz appear at some point, you just don’t know when. Morlet wavelets can be scaled to tell you precisely when 10kHz occurred in that window of time. “Finding out when a flautist plays a C sharp is the kind of thing you can do with a Morlet wavelet,” said Hayes. “You can set up the scale of a Morlet wavelet, so when you put a signal through, it only stands up when the flute plays a C sharp. They decompose the signal to when that condition is met, then suddenly turn on, no matter how much signalto-noise there is. And they have a nice property in that they’re perfectly time-aligned. So there might be latency, but the standing up and down of events is perfectly time-aligned.” And Hayes sees this as a key to understanding how our brains perceive sound. Cochlear implants work on a similar principle, it recognises that a certain frequency is happening at a certain time, and transmits a jolt of electricity into the basilar membrane. Because when you think about it, our basilar membranes are set up to decode sound in a similar way. Got a low frequency coming through?
Well, it’s only being picked up by the hairs right down at the skinny end of your cochlear spiral. Likewise, high frequencies are being decoded right at the throat of the horn. Hayes has a suspicion that the parts of the ear you can see, the pinna, is the wavelet encoder, which gives you the directionality as well the frequency. It explains why cochlear implants aren’t useful for spatial cues. But that’s just little more than a strong hunch at this stage. THE RESIDUAL BENEFITS
So, where does that put us? Well, Hayes sees that a quadratic residue diffusor, when correctly tuned, “produces a ball of scaled wavelets that when added together have a flat response, are perfectly diffuse, and have no statistics.” So from tests in anechoic chambers, if you shoot 6kHz at a QRD, you’ll get a nicely-distributed petal pattern. Do the same at 8kHz, and you’ll get a different petal pattern.” Basically, the QRD is de-composing the signal into scaled Morlet wavelets. So by firing your source into a tuned QRD, you’re de-composing the signal straight off the bat (across the QRD’s operating range). The result is supposed to be akin to an acoustic hologram, where the speaker satellites reconstruct the acoustics and depth of the original recording more accurately than a conventional nearfield set. Instead of wrestling with two sets of statistics — the set from the original acoustic plus the set from your listening environment — you only have one. The second benefit of this de-composition, according to Hayes, is that it theoretically has
SPATIAL RECALL In 2001 I reviewed a pair of monitors based on the same design principals as the Emergence AS8-Pro S system. The designer, Joseph Hayes, had spent quite some time setting the monitors and associated sub woofer in my studio, but in all honesty, this was hardly required as the Spatial One system (as it was then branded) sounded brilliant wherever you placed the units. The highly irregular design reminded me of a monitor I’d seen during the 1980s (the Omni F80), which as it transpired, was also designed by Mr Hayes. The monitors incorporated ‘CHAOS Speaker Technology’, whereby the upper frequency drivers were directed toward a diffuser just like the Emergence system. I was immediately hooked on the design and the stereo image. The Spatial One monitors ended up becoming my own, and to this day still function as my ‘every day monitors’. For all day listening and stereo imaging wherever you’re sitting, they are extremely un-fatiguing and a pleasurable listening experience. They’re the monitors I switch to to ‘recentre’ my hearing and sit between my near/mid field monitors and my extremely near-field Auratones. While I didn’t find the Emergence system as much to my liking as the Spatial Ones, the system does provide a very enjoyable, and rather wide, ‘sweet-spot’ and the same uncanny imaging.
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zero autocorrelation. There’s no really easy way to explain autocorrelation or its uses, except that it’s taking a signal and cross-correlating it with itself by pushing a copy in time to find similarities. Hayes believes: “we seem to have semi-objective proof now, that when something with zero autocorrelation reflects off a wall, there is immediately a reversal effect, and it self extinguishes. This is the anti-echoic effect of the speakers.” It’s completely different to how we usually think sound will respond when it hits a boundary — comb filters, ugly reflections, the lot. It’s a big claim, but the simplest experimental proof he could offer was to record a hand clap or impulse and play it back through the AS8 speakers into a room with a nasty acoustic. The diffused source from the speaker will have zero autocorrelation and self-extinguish at the boundaries, while the unrecorded hand clap will flutter as normal. SOUND OF TWO HANDS CLAPPING
After listening to these speakers in different environments for a while I decided to go ahead with the hand clap test. I took my dead hand clap recording into my eldest son’s room. He’s only a little tacker, so all his stuff is miniaturised and hasn’t quite grown into the size of his bedroom. It’s pretty square, has mostly hard surfaces, and flutters like a moth stuck in between your door and fly screen. I set up the AS8 Pro S system alongside a Dynaudio powered monitor and got to work. This is a hard test to get right. For one, a big part of the sound of a hand clap is the room reflections. So recording a dead clap through mic, preamp and interface, then playing it back through a speaker, is difficult to nail — it’s hard to know what you’re AT 44
trying to match exactly. Anyway, I pressed on and played a variety of material through both speaker sets, including different hand claps at different levels, synthesised dry hand claps, and other synthesised impulses to see what effects I could trigger in the room acoustic. Each time, I would set the levels of both speakers to read a target dB SPL at one metre. I was mostly concerned with hearing the difference in response between the two speakers — the AS8 Pro S, and the Dynaudio standing in as the conventional nearfield monitor — rather than trying to match the recorded hand clap to me physically clapping my hands together. I did notice a bit of difference in the reflections coming back from both sources. The Dynaudio was overall, more pronounced. But it definitely wasn’t the raging success I — must confess — was hoping for. It brought out a weakness in the design, that’s obvious to anyone with a passing knowledge of QRDs. That is, it’s pretty much impossible to build a completely broadband QRD. To get down to lower frequencies the wells have to be very deep, and the higher frequency cutoff is determined by the width of the wells — skinnier goes higher. So, even though the AS8 Pro S’s diffusor is tuned for the driver placement, its reach doesn’t extend down into the mid range and low frequencies below 1400Hz. You’re still at the mercy of your room acoustics past that point. And that’s what I was hearing in this experiment. The AS8 Pro S’s had some discernible effect at the top of the range, but the main issues in the room were happening below that range and could be heard no matter the source; physical hand clap or speaker. DSP DIGRESSIONS
To compensate for this lack of control in the middle and down low, the AS8 Pro S has a built-in DSP, which gives the user control over the voicing of both the satellites and sub. There’s a forum of users on the A3D Emergence website who’ve been playing around with different voicings, and the later DSP settings have become more balanced. You can also ping the room with a low-budget measurement mic and DSP, to give you a bit more of a personalised setting for your room’s acoustics. It’s impossible to talk about the voicing of the system, because not only is it constantly changing, it’s also user adjustable. With a Windows PC, you can get in and adjust the voicing with a whole suite of filters, and upload them to the DSP via USB. There isn’t an application for Mac OS at the moment, but there is a plan in the works to push all the DSP adjustments to a web browser environment, making it platform agnostic. THE HOLOGRAM EFFECT
It’s a difficult task trying to compare the AS8 Pro S system with a more conventional two- or threeway nearfield monitor. The sound stage it sets is just entirely different. While the hand clap test showed up the lack of diffusion below 1400Hz, the QRD creates a unique effect above that point. The effect is described as an acoustic hologram. But while it seems an apt description, my image of what a hologram looks like is coloured by early
sci-fi, where the hologram sits above a point, or projected from a point into a room. What the AS8 Pro S system does is surround itself with an image of the original recording’s space. So, things that are recessed in the sound field feel further back than on a standard stereo speaker configuration, and vocals also feel like they sit more in the picture, rather than in front. Overall, it made my typical nearfield setups feel more like they were presenting the music in line with an imaginary cord strung between the two monitors. It takes a bit of adjustment to mix on this system, especially if you’ve relied on the same monitoring system for a while. I made a few direct to stereo recordings in order to hear how the system replicated the space, and started to notice not only how deep the sound stage presented, but how far outside the speakers some sounds felt. I replayed Jeff Lang’s record through the system (AT Issue 102), which was mixed by placing prerecorded instrument performances in different speakers around an ABC live room, then rerecording it through a stereo ribbon and omni condenser. The result is a unique sense of space in itself, but through the AS8 Pro S system, it was like reconstructing the speakers in the room. FINISH UP
Now we’ve gone through the technical principles of the system, I guess I should actually make some conclusions about the product itself. My main problems with the device is that without having the Windows app open, there’s no feedback on the system’s settings. Because it has a built-in DAC, you can switch between multiple inputs on the supplied remote — analogue (stereo RCA or mini-jack), S/PDIF, TOSLink and USB. The problem I found was the lack of LED or LCD displaying the current input. I found myself scrolling through the inputs depending on where I had the system setup and what program I had running. Not knowing you’re on the right input and trying to troubleshoot other signal chain issues can be problematic. There are also volume controls on both the remote and sub unit, but neither give you an indication of the unit’s current level (bad news if you change inputs, go hunting for the new source and find the input level is a lot higher than the last one). Lucky the amp has a protection circuit in it to stop it overloading. The satellite shape is a bit hi-fi weird, but more a ‘form follows function’-type of arrangement. I can live with it. And there are constant refinements happening behind the scenes to increase the rigidity. Would I recommend it? Well, while it’s not ‘impulse buy’ cheap, at under $1000 for a flexible, unique system that will add a completely different perspective to your monitoring choices, it’s definitely worth trying to hear a system. At worst, you’ll have a really capable hi-fi to use around the house that will draw you back into the space of your favourite recordings. And though it’ll allow you to cut down on acoustic treatment in your home studio, I don’t think you’ll escape bass trapping and mid frequency diffusion for now.
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www.sounddistribution.com.au AT 45
REVIEW
iZOTOPE OZONE 6 ADVANCED Ozone’s interface has been reshuffled and a few things have gone missing. But Ozone Advanced users will have plenty of uses for the new Dynamic EQ.
NEED TO KNOW
Review: William Bowden
PRICE Ozone 6 (Boxed): $269 Ozone 6 (Serial): $249 Ozone 6 Advanced (Serial): $999 Ozone 6 Advanced upgrade: US$299 CONTACT Electric Factory: (03) 9474 1000 or sales@elfa.com.au
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PROS Dynamic EQ worthy new addition Standalone app & external plug-ins give flexibility Improvement in sound & workflow over V5 Trickle down features from V5 Advanced into V6 Standard
CONS V5 Reverb & Amount sliders gone missing Constant ‘all buttons on’ CPU Load
SUMMARY Ozone 6’s new look comes with flexibility — drag and drop module ordering and the ability to inject external plug-ins into the Ozone workflow. Some features have gone missing in the reshuffle, but the new Dynamic EQ module will have Advanced users contemplating the upgrade.
The Dynamic EQ is one of the hallmark features of Ozone 6. If, for example, you want to boost the midrange in a vocal, but don’t when the vocal gets loud — then voila: dynamic EQ is the answer.
As a company, iZotope has been making a nice little name for itself over the years. Its flagship mastering application, Ozone, amazed many people with its feature set and price when it was released; and each year or so has seen progressive upgrades and improvements. In case you haven’t been keeping abreast, Ozone is an ‘all-in-one’ software mastering solution comprising EQ, harmonics generation, image manipulation, compression, output limiter and dithering. A few years back iZotope began splitting products into Standard and Advanced versions, and I had the latter version of Ozone 6 on review. So what’s the difference between Standard and Advanced? Well, I’d advise going to iZotope’s website and look at the handy comparison chart as there are quite a few differences (apart from $750!), but I will quickly focus on a couple here. Firstly the good news: Ozone 6 Advanced has a new dynamic equaliser and I have to say it works rather well. I may be getting a little ahead of myself here but this is the big carrot and one of the main reasons to choose the deluxe version over its pared back sibling. It also might be a reason for some people to consider upgrading from Ozone 5, though more on that later. GREEN MEANS GO
So let’s start at the top then. Ozone 3, 4 and 5 all had similar looking interfaces and it was pretty easy to see the evolution progressing in terms of user interface and layout. Ozone 6 Advanced has a completely new grey look, layout, and in some areas, functionality. No more alien green, thank goodness. In use everything is very clearly labelled and whilst different to its forebears, I found it’s a piece of cake to get around. In terms of processing choices you have: EQ, Dynamic EQ, Maximiser, Imager, and Exciter. If you’re expecting to see the familiar Mastering Reverb — then sorry folks, it’s not here anymore, another slightly baffling move on iZotope’s part. You do however have the ability to customise the order of your signal chain; as iZotope has finally broken each processing unit into modules that
aren’t locked into a set cascade like they were in previous iterations. Swapping around modules is as easy as drag and drop, and new modules (up to a maximum of six) can be simply accessed from a palette. It’s also possible to do this in real time without a single glitch. Also, instead of just an iZotope module you can now insert third-party plug-ins. This is a great idea and being able to use Ozone in and around other plugs opens up quite a few possibilities sonically. It saves a bit of CPU power too, as before you had to open up multiple instances of Ozone to achieve a similar workflow. However, on the subject of CPU power there appears to be a major change in internal architecture. In previous Ozone releases the CPU load dropped or increased depending on the number of internal processors in use. Ozone 6 Advanced shows the same load regardless of whether all modules are in or out. This load is about the same as Ozone 5 with all processors turned on. Since we all value our CPU levels highly this does seems a trifle odd I must admit. Again, this is where the Advanced version’s single module plug-ins come in handy. As a reference, the solo Dynamic EQ plug-in loaded the CPU to about half the level of the full-blown Ozone package on my PC.
peak bell and proportional Q. Frequency range is 20Hz to 20kHz with linear phase and minimum phase available but only globally (sorry tweak-perband freaks). The compression section features a wide-ranging threshold, attack (0-100ms), release (0-1000ms) and the compressor has an inverse function to allow expansion of boosts or cuts — great for reclaiming some lost dynamics or really killing something! The boosts are limited to +15dB and the cuts to -30dB. Strangely though these values are only available if you type them in directly. If you just move the gain points in the graphic display then it’s limited to +5.4dB and -9dB — an oddly asymmetrical default range. Having said that, lifting out vocal regions, and giving guitars more shimmer or depth in MS was a joy with this equaliser. You really can make music sound better with this one module (or plug-in) alone. Per-band bypass makes life easy and having the ‘instant reset all bands to zero’ button was a nice touch too. Did I find any faults with this module? Not really, it’s very intuitive and sounds great. Does it face some stiff competition? Absolutely. Voxengo’s Gliss EQ springs to mind amongst others and it’s only $89. Melda Production’s MAutoDynamicEQ is a similar price and dare I mention Toneboosters’ TB Fix at just $20?
DYNAMIC ADDITION
TRICKLE DOWN
There’s a lot of minutiae in an update to an all-inone package like this. So I’ll focus on the new kid on the block, the Dynamic EQ. I’m quite familiar with my hardware dynamic EQs, namely the Tomo Audiolabs Lisa and the CLM Expounder. Essentially a dynamic equaliser allows you to add or subtract at a particular frequency, but modulates the amount of addition or subtraction based on a detected threshold (i.e. a compressor or expander). So if, for example, you want to boost the midrange in a vocal, but don’t when the vocal gets loud — then voila: dynamic EQ is the answer. Ozone 6 Advanced has a four-band dynamic EQ with a good number of shapes available per band: Baxandall bass and treble shelves, band shelf,
While the Dynamic EQ is an Advanced feature only, there are a couple of worthy features that have trickled down from the Advanced version of Ozone 5 into the Standard version of 6. EQ Matching lets you take a source piece of music, analyse and capture its frequency spectrum profile, and impose it on another track (or sound). It’s quite an interesting and educational journey seeing how you can bring one tonal flavour of a mix closer to another. There are parameters for smoothing and amount — which help you decide how far you should go in copying someone else’s tone. The Imager has a Stereoize function; which takes a mono source and applies some tricky processing to make it sound stereo. It can range from pretty AT 47
Ozone 6 lets you incorporate VST and AU plug-ins into its workflow, but there are already a lot of powerful tools in the package. Clockwise from top left: The Imager; the dynamics section; the exciter with new tube modes; and at the end of the chain, Izotope’s MBIT dither.
useful to mildly electronic and phasey sounding. I just have the basic version of Ozone 5 myself, and finding comparison specs for Ozone 5 Advanced on iZotope’s site is no longer possible — it’s like it never existed. For us mortals though Ozone 5 has very much existed and continues to endure. If Ozone 6 Advanced had been released without a forebear then this might be a fairly different review. As it is I found myself constantly comparing it to the previous release and often noticed things missing. Where’s the Reverb? I found this quite a useful tool and continue to do so. The Maximiser on Ozone 6 Advanced is missing a couple of features its predecessor carried, namely the Hard and Soft buttons, which have been replaced by more drop-down options under the IRC settings. A far more important omission in the new version are the Amount sliders. It might not sound like a big deal, but if everything is sitting really well and the client just wants a smidge more or a smidge less of everything, then a slider to vary the overall amount of processing takes a lot less time and is less prone to errors than going in and pulling it all down or up on individual bands. Try doing that on the multiband compressor and see how long it is before you start cursing. To be fair, the Maximiser in Ozone 6 Advanced seemed to sound better than the one in version five AT 48
— especially under heavy loads — though there wasn’t a huge difference in the ‘normal’ operating range. Similarly the new EQ seemed a bit more refined and subtle. Setting up what appeared to be the same values in Ozone 5 always resulted in a brasher sound, though that could usually be tamed by just turning it down a little. Finally, the new Exciter module definitely benefits from having two extra tube triode modes. SUMMING UP
So to sum up there are many things to like in Ozone 6 Advanced. Apart from the main additions, there are funny little hidden quirks like a parameter randomiser — the kind of thing more at home in a virtual synth for people who need an oblique strategy every so often. You also can use Ozone 6 as a standalone app and have basic fade in and fade out control of a wave-file appearing at the top of the window (or several wave-files which you can tab between). It’s a useful mode of operation that draws your focus to the mastering process, and you’re not short-changed, especially now you can integrate other plug-ins directly into the Ozone workflow. There’s also a function which auto detects various changes in song structure and maps these as regions you can jump to — works like a charm. Overall, it seems Ozone hasn’t really made the same kind of quantum leap in features it did when
going from version four to five, and the market is extremely competitive now. The biggest changes are in the things users might not place as much value on, like GUI enhancements and workflow features. And although the GUI’s new look is inline with more celebrated modern plug-in designs, I found the graphical feedback a little rudimentary for a flagship product. It’s also not inexpensive at just under a thousand bucks. Will the basic version of Ozone 6 sell as well as Ozone 5 did? I would say it’s unlikely, it’s a hard act to follow. As for my advice — try it in demo mode and see what you think. There are certainly sonic differences between 5 and 6 which may sway some people, and the upgrade pricing is worth a serious look for the Dynamic EQ. On that note, if you’re in Australia and considering buying Ozone 6 or an upgrade, make sure you head along to a local dealer. Electric Factory has pulled out all stops and ensured that it’s cheaper to buy local than online. Nice one! Ozone 6 Advanced faces very stiff competition as many of its features are available from different and cheaper software houses, but once again iZotope provides a tried and tested all-in-one approach. Whether such an approach is the right one only time will tell, and alas, I am out of it folks.
“ Spatial detail without loosing the big picture.” Dr. Antti Sakari Saario Head of Music, Falmouth University
“beautifully honest.”
For all demo and sales enquiries, contact: Federal Audio sales@federalaudio.com.au AT 49
REVIEW
RODE STEREO VIDEOMIC X It’s still got a 3.5mm output, but that’s the only concession in this otherwise completely professional new addition to Rode’s Videomic line. Review: Mark Davie
NEED TO KNOW
Though mics from Rode’s Videomic series have found their way into most video camera bags, the third addition to its Stereo Videomic line, the X, shows a real commitment to professional folk. For one, it has balanced stereo outputs on mini-XLR connectors that can interface with professional gear, and also be used to feed the mic phantom power. When there’s no phantom power on hand, or you’re using the stereo 3.5mm output jack instead, power is supplied by a 9V battery, securely housed behind a metal door. The two 1/2-inch, externally-biased, true condenser cardioid capsules are acoustically matched. And unlike some other stereo mics, including the ones that come attached to a lot of portable recorders, it doesn’t exhibit any phase issues — it’s real broadcast-
PRICE $899 CONTACT Rode Microphones: (02) 9648 5855 or sales@rode.com
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PROS True condenser quality sound Excellent build quality Popshield & wind shield included Pro I/O Great options
CONS None
quality sound, mated to Rode’s incredibly quiet electronics. It’s not Rode’s quietest mic, but its Equivalent Noise Level of 12dBA is very good. It can also handle a maximum 143dB SPL, which is very healthy, and necessary when you’re out in the field surrounded by sources with an unknown dynamic range and output. The onboard preamp also has a -10dB cut and a clean +20dB level boost to accommodate different sources and recording devices. The level select button is one of four on the back of the unit housing. The others are power on/off, high-pass filter and high shelf. The buttons toggle through each setting, with mini LEDs denoting the selected position. It’s a great system, and better than the traditional sliding switches.
SUMMARY Rode’s Stereo Videomic X is a jump up from what we’re used to seeing in the Videomic line. Professional features like an aluminium chassis, Rycote Lyre suspension, balanced stereo mini-XLR outputs with phantom power, and included pop and wind shields all complement the sound of the acousticallymatched, 1/2 -inch true condenser capsules inside.
BUILD SHOCK
Rode’s slightly older Stereo Videomic Pro houses its X-Y capsules in a cage, with the whole unit, including preamplifier, suspended by rubber bands. The X version shows how serious Rode is taking these sound-for-video accessories. The all-aluminium body is ruggedly built, and has an upward tilt to accommodate both capsules in a perfectly stacked, 90° X-Y pattern without any interference between them. Both capsules have individual shockmounts in a mini-Rycote Lyre setup made of a single piece of thermoplastic. They’re much hardier than any rubber band suspension system, and work far better too. Plus, the suspension mounts clip into place rather than being glued, suggesting they could be replaced if damaged down the track. The Stereo Videomic X does come with a 10-year warranty too, so you’re well covered in any case. The whole unit weighs 300g, which is less than a quarter of the weight of typical small DSLR and zoom combo. Everything adds up, but for the solid construction and level of audio quality you get from the Stereo Videomic X, it’s not too much to bear. POP ON A SHIELD
The naked capsules are much more resistant to wind noise than the Zoom H6’s, for instance. And once you put on the pop shield, it cuts out any short gusts quite comfortably. It’s made of durable foam that won’t rip, and has a layer of rubber hexagonal webbing that extends around the shield for extra protection. The rubber-coated, plastic mouth swallows up the X comfortably, with the inside of the foam cut out perfectly to accommodate the exact shape of the capsules. It clips on with the grip of grandma’s teeth; firm, but no trouble prying off. With the dead cat wind shield on, I could swing the mic right past a fan blasting on high, without any adverse affect. You’d have to have be in the midst of a gale to have any issue. To compensate for any loss of high end with the wind shield on, Rode has incorporated a switchable +6dB high-shelf filter. There’s also a two-stage high-pass filter, with what sounds like quite a steep dB/oct slope. It’s useful for chopping out nearby low rumbling noises — road noises, air conditioners, the usual occupational hazards. It’s not a foolproof solution, but necessary to have. The higher 150Hz setting does cut out a chunk of the low end in a male voice, so just watch that depending on the talent in front of the camera. As far as cons go, you could note the lack of mini-XLR cables. But that’s like looking a gift horse in the mouth. The 3.5mm spiral connector will suffice for most users. And it’s a great-sounding stereo mic, worth the money, especially since you get every other custom accessory you need in the box.
Distributed by
40 Kent Road • Mascot NSW 2020 Ph: (02) 9582 0909 • Fax: (02) 9582 0999 www.jands.com.au • info@jands.com.au
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REVIEW
PRESONUS AUDIOBOX iTWO iPad Audio Interface Wirelessly transferring your demos straight into your DAW is a neat trick only Presonus can do.
NEED TO KNOW
Review: Mark Davie
PRICE $249 CONTACT National Audio Systems: (03) 8756 2600 or sales@nationalaudio.com.au
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PROS Wirelessly transfer demos to your DAW Class Compliant USB for multi-device connectivity MIDI input for iPad synth control
CONS No effects in Capture app Trickle charge only props up battery life
SUMMARY Presonus’ Audiobox iTwo makes embellishing demos a cinch. Record on your couch and wirelessly transfer your session to your computer for fast-tracking ideas.
I’ve really been getting into the spirit of it; lounging back into the studio couchback, strumming a guitar, and warbling into a Shure SM7. Tethered to the tail end of my cables has been a relatively portable two-piece setup, comprising a Presonus AudioBox iTwo interface hooked up to my iPad — it’s demo nirvana. The real issue with demos is not finding some handy portable device to knock them out on, it’s in transforming those quick strokes of inspiration into a finely rendered production. Jumping between a portable recorder and your ‘serious’ recording system can be as messy as trying to mix watercolours with oil paints. There are some nice solutions to this predicament — Garageband and Logic speak kindly to one another, and AAF Import functions tend to bridge the gap between sessions — but Presonus’ wireless transfer of sessions between your iPad and laptop or desktop is very neat indeed. JUMP THE GAP
But we’re jumping ahead of ourselves. Firstly, lets’s get back to the hardware; the Audiobox iTwo. By taking the bones of its already successful two-channel Audiobox range and adding an iOS connection, Presonus has doubled its flexibility. There’s not much new here with the iTwo. Two combo, auto-sensing mic/line/instrument inputs on the front panel, with a button to switch between the latter two. Individual gain control pots give you 52dB of gain, and a single button switches on 48V phantom power for both channels. A big master volume pot controls the main stereo output on 1/4-inch TRS, and there’s a single headphone output with its own volume control too. Right in the middle of the unit is the mix control typical of small interfaces. Forget buffer settings, just dial in a comfortable balance between computer playback and input source for zero latency monitoring. Slightly odd, but welcome, are the MIDI DIN ports on the rear. Odd, because most small twochannel interfaces don’t sport MIDI — especially not on DINs, given most devices transmit MIDI over USB now. But welcome, because the iTwo can double as a quality audio output for your iPad synths. CAPTURE IT
Back to the software side. The key to Presonus’ little magic teleportation act is the combination of its Capture iPad app and Studio One computer DAW. You can download Capture Duo — the twotrack, 24-bit/96k-capable iPad app — for nix on the App Store. Which is neat for podcasters and simple demos. But if you want to break into more tracks, you can pay $12 to expand Capture’s 96k track count to 32, turn to the included Garageband (limited to 44.1k), or shell out for something like Auria. Capture Duo is suitably easy to use, and well set out. The touch buttons can take a fat finger, and it automatically snaps to time or meter grids, making it useful for quick singer/songwriter-y demos or podcasts. You can trim, split and duplicate regions,
easily undo or redo, set levels and pan, and create loop points for nailing that part. One nifty feature is the ability to easily change whether a track is mono or stereo without diving into menus or re-creating the track — a nice flow-over from Studio One. And there’s also a button that reveals how much recording time you have left on your iPad. There’s no effects, which is okay, but it would have been nice to see reverb and a guitar amp emulator onboard — handy inclusions for singer/ songwriter-y types. There’s also no drum machine or inter-app audio capability, so if you want to build relatively fleshed out demos you’ll have to turn to another app. But the real benefit of using Capture is that ability to wirelessly transfer your sessions straight into a copy of Presonus Studio One to polish them off. Pretty neat, and loads better than having to plug your iPad in, navigate to the app’s data in iTunes and manually save and open it. All you need to do is make sure both devices are logged into the same wireless network, and Studio One is open on your computer. Hit the Share button on your iPad, select the computer you want to share it with and, in a jiff, the session loads up as advertised in Studio One. Once it’s in you can immediately start getting to work fleshing out that masterpiece, no cables or iTunes navigating necessary. It saves a lot of time and it’s really that easy, just make sure everything is up to date and you have the latest iOS version. A download code for Studio One Artist is also included with the iTwo, which at $135 value is nothing to sneeze at. It’s one step above the free version of Studio One, and has most of the features of the bigger boys, including 28 native Presonus effects. Its only major drawback is the inability to use external plug-ins and Rewire in other software. But Presonus offers upgrades at decent prices if you want to move up the Studio One food chain. CABLING UP
One of the Audiobox’s selling points is that it can be USB bus-powered. But unfortunately, an iPad doesn’t put out enough to drive itself and the interface. So while not having to plug in iPad power would have made this a neat mobile package, you still have to. On the upside, the only accoutrements required for power are an extra USB cable and the charging pack that comes with your iPad — no extra powerpack to carry around. On the downside, this configuration doesn’t fully charge your iPad while recording. Though the iTwo does pass on any leftover power, trickle charging your iPad to keep it going. It took roughly 10 minutes of single-track recording to decrease the iPad battery by one percent. At that rate, you could record for over 16 hours straight. And recording multiple tracks didn’t seem to deplete the battery by much greater measure. It makes a great podcasting setup, especially if you already carry around an iPad anyway.
“
The real benefit of using Capture is the ability to wirelessly transfer your sessions straight into a copy of Studio One
”
IN USE
The iTwo doesn’t sport the XMAX preamps of Presonus’ upper range, but they’re still a class A design, and have a suitable gain range for most mics and sources. There’s no pads or phase flip, but you rarely see that in this level of interfaces. The outputs are really clear, with plenty of gain on the headphone amp. The preamps are good for this level; recording bass through the DI input retained plenty of low end, and guitars were fine too. Even absolutely cranking the gain to max to record a falsetto part on the low-output SM7, there was hardly any noticeable noise — quite impressive actually. Of course, this isn’t exclusively an iPad interface, you can use it with any computer as a handy two-channel interface too. And because it’s classcompliant, it’ll work with your iPhone too, just without the Capture app and its wireless transfer. There’s also an iOne interface, which cuts out the MIDI, and drops the channel count to one. For the price, the iTwo was always going to be good value as a multi-purpose two-channel interface. But with onboard MIDI, it makes itself useful for live performance too, and the wireless transfer function is more than just a gimmick. If you subtract the $135 value of the included Studio One software, you’re paying barely $100 for a really useful piece of kit. AT 53
REVIEW
WAVEDNA LIQUID RHYTHM Liquid Rhythm breaks beats down to the molecular level for deep-tissue manipulation. Review: Jason Hearn
1
NEED TO KNOW
Although I’ve developed dexterous finger drumming skills over the years, beat composition tools have become a gateway to more exotic grooves I’d never play habitually. WaveDNA’s Liquid Rhythm is a rhythm sequencer that — rather than plotting in individual notes into a piano roll or sequencer — cheats for you and puts in note clusters. It’s all based around WaveDNA’s Molecule system, which presents MIDI in a whole new way. Liquid Rhythm can be invoked as a standalone instrument or DAW plug-in (AU/VST/RTAS). Both modes include a built-in drum sample player, but when opened as a Max4Live device it only displays the sequencer, relying on the user to insert a Drum Rack after it to provide sounds — a preferable option for anyone using their own drum kits. It’s clear the sample player is not the focus here. A mere four drum kits are preloaded (expandable to 12
PRICE Liquid Rhythm: US$129 Liquid Rhythm Intro: US$49 CONTACT www.wavedna.com
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PROS New, deep way of looking at beats. Inspired randomisation, maps and groove tools. Seamlessly integrates with Ableton Live.
WAVEDNA MOLECULE You build a Molecule from the bottom up. The bottom row is the BarForm, which breaks up the bar into 1/8th notes, grouped into either clumps of two (blue) or three (red). Above that are the BeatForms, which are smaller slices within those 1/8th divisions — they can be a simple 1/8th-long slice (purple), two 1/16th slices, triplets, or even up to six slices
CONS Tempos outside of 4/4 take a bit of a workaround.
per division for those super-fast glitches. With the timing in place, grey diamonds above each BarForm slice represent note events, and the grey bar above each note denotes its velocity. The shade of each colour is another layer of grouping, which allows you to tailor the velocities of the notes in each shade, or the groove offset. It’s deep, man!
SUMMARY WaveDNA’s Liquid Rhythm dissects beats to give you a whole new way of looking at them. It’s an inspired tool for production in any DAW, but Ableton Live users will really benefit from the Max4Live and Push integration — especially in live performance.
2
3
4
1 Liquid Rhythm hosted as a Max4Live device inside Ableton Live allowing dynamic editing of Live’s MIDI Clips. 2 The Instrument Editor allows you to create velocityswitched, multi-sampled instruments. 3
Liquid Rhythm as a VST plug-in.
The circular BarForm/BeatForm Maps with the BeatWeaver on the right. 4
MAX4LIVE HAT TRICK
via free download), built from 200+ instruments comprising velocity switched multi-samples. The Library also contains 769 loops, which may serve as an easy starting point for beginners. It’s possible to build kits from scratch using your own drum samples, however don’t expect more than a basic multi-sample player since there are no synthesis functions. Liquid Rhythm is best considered a specialised sequencer for the other drum instruments in your DAW. BUILDING BEATS
When opened as a plug-in, Liquid Rhythm’s Library appears on the left; the Arranger, Mixing panel, and Molecule Tools down the centre; and the Beat Builder to the right. After loading a kit from the Library, you create beats by clicking on an empty measure on the Arranger canvas and selecting a BarForm from the Beat Builder. The selected instrument type determines the initial palette of suggested BarForms. Once you’ve selected a ballpark BarForm, you can use the BeatForm Sequencer to add, subtract or
replace note clusters in the current measure. Delving further into this new mode of beatmaking, you can also use the BeatWeaver and circular BarForm/BeatForm Maps to construct beats. The BeatWeaver basically lets you dial in (with a pot) the complexity of each BeatForm, while keeping the others in the measure locked in. And the Maps present palettes of 256 BeatForms/ BarForms, organised by increasing complexity, that may be dragged onto the canvas. If your beat’s still lacking magic, Molecule Tools can add randomisation and dynamic accent/groove functions via the Randomizer, GrooveMover, Accent Mods, BeatForm Palette, BeatForm Tumbler, and BeatForm Shifter. All the parameters can be MIDI mapped for mouse-free interaction. Of course, if you want to manually sculpt your beats, you can engage Drawing or Painting mode to insert notes or BeatForms directly on the canvas. This can aid in finessing the results of randomlyspawned patterns. And a drag’n’drop style MIDI export function is provided for arranging patterns you create within DAW tracks.
I really started to hit my stride with Liquid Rhythm when using it as a Max4Live MIDI effect. It sits in series before a Drum Rack, which means events within the currently selected clip (including kit piece names from the Drum Rack) appear in Liquid Rhythm’s Arrangement canvas. Any changes you make in Liquid Rhythm are dynamically updated in the Live clip. Better still, you can apply Live’s Groove Templates to clips and hear the results in real-time. Used in this manner you could consider it to be a rhythm-dedicated ‘inspector’ for Live’s MIDI Clips allowing you to add subtle fills and variations to an existing drum track. It may also be configured for use with non-Drum Rack-based clips — i.e. other drum synths or melodic parts can join the party. PUSH TO DEVICE
Liquid Rhythm can be deeply integrated with a selection of control surfaces including Ableton’s Push, Livid Instrument’s Base II and Smithson Martin’s Emulator Elite touch surface. With these controllers you gain hands-on control of the BeatForm step sequencer, arrangement, humanisation and groove functions that could spice up live performances. For those indulging in rhythmically complicated genres such as Dubstep, Drum’n’Bass, Future Bass and Glitch Hop, this instrument provides fertile ground for cultivating intricate beats with little effort, while maintaining the hallmarks of your preferred sound design workflow. My only criticism would be that it requires workarounds to operate in time signatures outside of 4/4. Although it takes some time to master its voodoo, Liquid Rhythm may hold the key to busting a beat maker’s block! AT 55
REGULARS
LAST WORD with
Bill Putnam Jnr, Founder Universal Audio
Following in the footsteps of his pioneering father, Bill has built a business that provides the highest quality analogue gear for digital people, and digital processing that even analogue die-hards can’t fault.
I don’t think I ever consciously decided to ‘follow in my father’s footsteps’. It was the result of a sequence of things. The first thing my dad did was infect me with a love of music. Even before I really understood what the audio business or recording business was or what recording studios were, he would take me to go see great music. The first live show he took me to was Duke Ellington. As a kid going into a concert hall, there was something in the atmosphere that was just electric — I could almost touch it. My dad also introduced me to good sound. Going down to studios, and pulling up the mics in the live room and knowing what that sounds like. I wasn’t there specifically for an education, but it just happens as a young kid if you’re around great music and great sound. My dad bought me a radio kit for my 10th birthday and a toolbox, which I still have, and I wrote my name Billy Putnam on it, which is still somewhat legible. It had a soldering iron, a wirecutter, a couple of screwdrivers and a kit called Globe Patrol, which was a radio kit from Radio Shack. We built a shortwave receiver. I’d climb up a tree and carry this yellow antenna wire. We turned it on and I heard this English voice. It was the BBC and it was like magic — technology was absolute magic. The best advice my dad gave me was to find a profession you’d probably do regardless of whether you were paid or not. He knew the importance of loving what you do. So much so, it was less advice and more a dictate. Like, this is really what you should do: find that thing and do it — it’s totally possible. I went to college to study physics, and finally concluded: why am I studying physics? I really want to be an electrical engineer and I really want to do signal processing. I was into my music and I decided I wanted to discover something new about how people hear music and how people enjoy it and relate to it. I made that decision after my dad had passed away. My post-grad studies in signal processing were at Stanford — a Ph.D in Electrical Engineering with a wonderful group called CCRMA (we pronounced it ‘karma’) or Center for Computer Research in Music & Acoustics. THESE DAYS QUITE A FEW ACADEMIC CENTRES TAKE AUDIO AND MUSIC SERIOUSLY BUT BACK THEN IN THE MID-’90S IT WAS MOSTLY TAKEN AS A JOKE.
CCRMA was one of the few places that did take it seriously. And so I was lucky to be at a place that really got into the technology of music.
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Stanford is in the Silicon Valley. All my friends wanted to be entrepreneurial and I got that bug. So there was a point when I was deciding if I would spend my life being an academic and teach, or being an entrepreneur and have a business. Finally, I thought, ‘geez I can have it all, I’m gonna go down the business path and I’m gonna find opportunities to teach’. So that was the conscious decision at that point. Life in business is all about learning from your mistakes and I’ve made every mistake. First lesson: every act of creation has to start with the suspension of disbelief. You have to let an idea live and breathe before you start to poke holes in it; let it grow legs before you think of all the reasons not to do it. ’Cos if you’re a sane person, you would easily and justifiably talk yourself out of any ambitious act. If you knew all the hurdles you’d face beforehand… you wouldn’t do it. Once the idea has a life of its own, then it’s time to shoot holes in it — look at those vulnerabilities and try to fix them. But if you shoot holes at the idea too soon, you’ll never do it. You’ll get talked out of it. My brother Jim and I had very different opinions about bringing UA into the digital realm. It was the late ’90s and I was very excited about applying the DSP knowledge I’d acquired at college. Digital audio was in its infancy but it represented a world of opportunity. Meanwhile, my brother thought there was no reason to use anything but a great twoinch tape machine, a nice console, and a bunch of outboard equipment. What the heck are you doing? Digital?! I WAS CONVINCED A LOT OF WHAT PEOPLE WERE SAYING ABOUT THE ANALOGUE WORLD WAS REAL, BUT I WAS ALSO CONVINCED IT DIDN’T HAVE TO BE THAT WAY. I was learning a lot of new techniques
and I was determined to apply them to audio processing in such a way that we could make digital capture all the great aspects of analogue. That was the idea in a nutshell. That was the original vision for Universal Audio. I’m really lucky to have my musical skills. There’s so many great musicians here at Universal Audio. There’s so many great musicians using our equipment. And I still get a vicarious thrill, making something that matters to someone else’s musical process. I love talking about gear. I do get lost in it. I spend more time talking and thinking about gear than sitting in front of a guitar. But my hope, both personally and for our customers, is we have the gear that allows us to spend more time in the music — allows us to take that idea, that vague concept in our head, and get it out into the world.
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