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ISSUE 85 AU $7.95(inc gst) NZ $10.95 (inc gst) File Under ‘Music’


SHURE BETA MICROPHONES Optimised for high sound pressure levels, extremely low handling noise and higher gain-before-feedback. Accurate, focused polar patterns. ns. Virtually indestructible. Precision control to practically eliminate bleed. Because thee sound should be as clear and inspired as the message.

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Kenton Forsythe

on the Charge of Innovation in the 1st Degree Over his 40-year career, this serial innovator has perpetrated such breakthroughs as the BH212 horn-loaded low frequency module, the KF900 Phased Point Source Array, EAW Resolution software and any number of real-world applications of the aforementioned violations of the status quo. He is likely armed with portable audio test and modeling software and should be considered dangerous to any low-quality sound reinforcement installation or poorly deployed concert touring sound system.

Wanted for Questioning: The Collaborators Craig Janssen Lead collaborator on original MH Series for stadium and large house of worship uses. Helped Forsythe overcome problems of LF pattern control without massive horns. Also responsible for early application work at Anaheim Stadium.

John Lyons Primarily responsible for Forsythe and EAW’s invasion and domination of Dance Nation. Known to frequent nightclubs of his own design, Lyons co-designed the Avalon Series of loudspeaker systems, including the notorious DCS2 dual 12-in, horn-loaded subwoofer.

Dave Robb Spent early days in concert touring, helping Forsythe implement MR102 and BH215 systems. Later helped Forsythe develop a range of installationspecific loudspeaker systems, primarily for performing arts applications.

David Gunness Directly responsible for Phased Point Source Technology, KF900 Series and related global outbreak of KF900 installations. One-time EAW gang member now leader of his own enterprise.

Ron Baker Played a key role in the widespread application of Forsythe innovations, including dozens of very large stadiums, arenas and houses of worship. Known associates include Jack Wrightson, Gary White, Tom Falgien, Kevin Day and Mark Graham.

Jeff Rocha Inculcated in the EAW gang at a young age, Rocha has risen to be head of the enterprise. Known to be responsible for KF750 Series and KF760 Series line array systems.

Production Audio Services (New Zealand) P. +64 (0) 9272 8041

Nathan Butler Newest of Forsythe’s collaborators, controls dayto-day operations at secret lab location known only as The Pit. Responsible for AX Series, QX Series, DSA Series, KF740 line array system and NT power and processing. Known to be planning further innovations.

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Top-selling Loudspeaker Line: K Series

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©2011 QSC Audio Products, LLC. All rights reserved. QSC and the QSC logo are registered trademarks of QSC Audio Products, LLC in the U.S. Patent and Trademark office and other countries. MI SalesTrak is a registered trademark of Marketing Information Services, Inc. in the US Patent office. AT 4

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Yamaha Commercial Audio provides design services, EASE ™ modelling, programming, commissioning and training to make your PA investment the best possible value for money. Yamaha integrated PA systems take care of your entire signal chain, from input to output, giving you perfect integration and an unbeatable one-supplier advantage. Contact Yamaha Commercial Audio on 03 9693 5272 or for more information and make your next installation the best it can be. Follow us on Twitter - Yamaha_CA_Aust Find us on LinkedIn - Yamaha Commercial Audio Australia AT 5

Editorial Director Christopher Holder Publisher Philip Spencer Graphic Design Leigh Ericksen Art Director Dominic Carey


Advertising Philip Spencer


Accounts Manager JenTemm Circulation Manager Miriam Mulcahy Proof Reading Calum Orr

The universe of iOS and Android apps is immense. But back when the iPhone was launched the galaxies of Business, Creative, Games, Lifestyle, Finance apps were only beginning to coalesce… and the world of Mobile Audio apps was like a deserted outpost on the ‘Outer Rim’. While I’m trawling out Star Wars allusions, I seem to recall that early iPhone adopters spent quite a bit of the time bragging about their iSabre – with choices of Darth Maul red or Skywalker blue, and ‘authentic’ light sabre battle noises. There, of course, were the early iOS games as well, which were about as sophisticated as Galaga. It took developers time to recognise how powerful/ pervasive the iOS platform would be and, of course, anything half decent takes some time to code. Fast forward to 2011 and there’s not a week that goes by without a nifty new mobile audio app hitting the market. Or, a nifty piece of audio hardware, that allows you to interface with your iThing. Up to this point, most apps have been for ‘noodling’ – self contained, idea incubators… get some funky loops happening, strum an iPhone ‘guitar’, record four tracks of audio for those ‘jam in the hotel room’ moments. Gradually, though, the apps have become more sophisticated and provide a genuine performance alternative to a computer, synthesiser, or hardware controller. Moreover, developers are now taking advantage of the extra screen real estate afforded by the iPad (not to mention the touch interface) to release apps that are way more suited to a mobile device than a laptop or a desktop computer. Take Moog’s Animoog synth as an example. It’s brilliant – sounds great, and the touchscreen iOS interface provides a better, more interactive, richer experience. No wonder Animoog was the App Store’s top performer for a while there. And this is the direction we’re headed. Mobile devices are AT 6

great for performance, and developers will be rethinking their GUIs as a consequence – rather than simply offering cut-down retreads of a plug-in they might have laying about. I’ve no doubt we’ll soon be hearing about iPad Orchestras – groups of performers, using the iPad to cue loops, play solos, control effects and visuals… Actually, I should jump onto Google, no doubt there’s a Japanese avant garde performance art troupe which has toured the world twice doing exactly that. The point is, it’s still early days for iOS and Android mobile audio apps, but AT wants to hear how you’re using your smartphone or tablet. We’re putting together a Mobile Audio App Special next issue and you’re invited to get involved. What’s your favourite mobile audio app? How has it changed your working methods? How have you integrated it into your current setup (if at all)?

Regular Contributors Martin Walker Michael Stavrou Calum Orr Paul Tingen Graeme Hague Paul McKercher Hugh Covill Adam McElnea Greg Walker William Bowden Greg Simmons Rob Squire Robin Gist Michael Carpenter Mark Woods Andrew Bencina Mark Bassett Chris Vallejo James Wilkinson Gareth Stuckey James Roche Distribution by: Network Distribution Company. 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. E: W:

Feel free to email me with as little or as much as you like. The best contributions will all receive a Tascam iXZ portable iThing interface from our friends at CMI Music & Audio. Consider it a bribe to encourage you to give this a little consideration. Andy Stewart’s not writing this Ed Space… you’ve probably figured that out already! After eight years in the Editor’s chair Andy makes way for a new editor in the new year – Mark Davie. A big thanks to Andy for bringing his insights, skill and great audio knowledge to the role over the years. Have a great Christmas – whether the period is a time of R&R or the busiest of year for you, we wish you all the very best. See you in 2012! Christopher Holder, Editorial Director

All material in this magazine is copyright © 2011 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. 1/12/2011.

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Readers Letters. NEWS

News and new product information.






Medici Studios near Geelong is making a name for itself in Japan for its Surf Rock! We talk to house engineer Michael Stangel about the Medici Method.


Around the studio traps, featuring Fishtank Studios. PC & MAC AUDIO

Martin Walker re-imagines Windows 8, while Brad Watts goes to sleep… or hibernates… ACTIVE PICKUP LINES

‘Are you an active or passive mix device?’ asks Graeme Hague.







FOH Engineer Brent Rawlings has been with Kings of Leon since Day 1. He tells us how he maintains a balance of power and control. This Christmas’s big videogame release has a big soundtrack. We learn how it was recorded and assembled.



We spend time in master mix engineer, Mark ‘Spike’ Stent’s LA studio to discuss the mixing of Coldplay’s latest masterstroke.





Stav enrols us in a crash course of Stavrophonics. Test yourself, and see how you stack up. DIs: YOU GET OUT WHAT YOU PUT IN

Trevor Cronin tackles the unglamorous subject of direct injection boxes. You’ll be glad he did. KEEP IT CLEAN

James Roche tells us how a suite of noise reduction and restoration plug-ins can save your skin.




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YOUR WORD Readers’ Letters


RE: SO YOU WANT THE SONG IN EIGHT HOURS, ISSUE 83: I picked up my copy of AT83 from the newsagents today. I will subscribe because I am feeling lucky about those ADAM monitors [subscription prize]. On a serious note, though, it contains what I would argue is the most inspirational article I have read in my days of reading this mag. The no nonsense approach, and work ethic of this guy [Blair Joscelyne] are something that everyone, regardless of what business you are in, should aspire to. What I took away from it was: know your tools regardless of what brand/price they are, know what you are trying to record and bloody well get it down and commit to it. It is the way I have started to do my stuff after a long period of accumulating gear and trying every plug-in under the sun. This article has really spurred me on an I will be using some of the ideas contained in it. Thanks for a great article Blair and AT mag. via email –– MUSIC TEACHER: GEAR & BEST PRACTICE?

I just picked up Issue 83 again in a moment of peace in the middle of a very busy school term (I’m a high school music teacher) and I thought: I have a couple of articles I wouldn’t mind seeing in the next few issues of AT. So here is my wish list: As a secondary music teacher, my needs are a little different to your average project studio/ recording musician/large boutique gear guy. While I have a fascination with technology myself, I spend most of my time considering the role that technology plays in the classroom – in particular, in regards to the general music education of high school students. It struck me as a little unusual, when I moonlighted for a brief period teaching a ‘Film, Television and New Media’ unit, that in that subject there is a whole criteria related to the use and implementation of technology. In music, the closest area that we have is Composition. However, this is still viewed primarily in regards to traditional compositional methodologies – not the use of technology. Certainly, in this regard, there is a lot of latitude for an individual teacher to focus more or less on the use of audio technology to capture the creative idea (be it through a traditional score, or through a sound canvas like ProTools or Logic). In this respect, I believe that music education should be moving in the direction of audio technology. As I have learned through the last seven or so years that I have subscribed to AT, irrespective of genre or style, music is best translated through the use of recording

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technology. This means that the better equipped and informed music teachers are, the better the outcomes for future musicians. I find this is particularly relevant given the shift to online marketing and distribution (through iTunes and Soundcloud) and the prevalence and access that students have to an endless list of cheap tools and equipment. Anyway, I digress, I was wondering if AT has any insight into ‘best practice’ approaches in high school education? For example, best software platform, best integration of technology in the classroom, best examples of classroom setups, best way to integrate recording technology education and practice, with existing curriculum requirements? Perhaps a regular education spot? My final request is this: what is the best laptop bag for music teachers? A review on the options would be good. Standard laptop bags just don’t cut it for me. I’m currently in the market, and the best options I have found at this point are: the Fusion F1 Workstation Backpack or the Mono DJ backpack. Any suggestions? I teach in Ipswich, QLD, and use ProTools, Logic, and Sibelius. I have moved towards Logic due to its site license price point (really cheap for our 60 Macs), but want to keep ProTools (v8) as an option for serious sound guys – haven’t decided to upgrade to 10 yet. Chris Hollier Chris has set up a topic on the AT forum that addresses his questions. Anyone who’d like to offer an insight would be most welcome to contribute. – Ed. –– G5 & ADEQUATE ‘TOOLS

RE: ED SPACE ISSUE 84: I’ve been an avid (no pun intended) reader of Audio Technology for many years and your latest editorial struck a chord with me as I went through the same drama 6 months ago (with my G5 – not my dog, which is actually a cat). After 6 or 7 years of very faithful service looking after my ProTools HD3 system, my G5 decided it was time for retirement and I too was not pleased to learn that the new Mac Pro does not support PCI cards. As my G5 failed to the point where it was useless, I even purchased a second hand G5 just to keep me going for a few months while I sorted out alternatives. The Avid upgrade path to PCIe cards is rather costly for a small studio, so I opted for a wiz-bang Mac Pro with dual 2.66GHz six-core processors and an upgrade to ProTools HD Native. It’s pretty darn good – no latency issues, as many tracks as you would want and, so far, no problem

with lots of plug-ins. It’s actually better than my old HD system! Good luck with the upgrade – and with the new seats (leather seats in a Mazda Bravo?), and thanks for a great magazine. Jack Setton ––

Know what you are trying to record and bloody well get it down and commit to it

RE: ED SPACE ISSUE 84: I made the leap to HD a couple of years ago, spending $9k on a second-hand HD2 rig which lives in my G5 of around 2005 vintage. It’s a f**king excellent system – bless its soul – and it has kept up with my needs very well. My number one reason for upgrading from LE was to get delay compensation, everything else I use is pretty much the same between HD and LE. Delay compensation was completely necessary, it seems dodgy that it wasn’t included with LE. I see phase as the single most powerful tool in audio, and LE made it completely un-controllable. So my $9000 purchase (total bargain at the time) provided me with an obsolete system only a few months later when PT9 was released… I wasn’t happy. I have looked at everything you touched on in your editorial (regarding upgrading), and the writing is on the wall: if I am to keep with the times and make the most of new features, I have to replace the whole system. So my gripe with Avid is this: in releasing PT9 with delay compensation and other HD features, you have de-valued every existing HD system on the planet to virtually zero, but have not offered much of an incentive to continue buying your product. Avid, at least make it easy for me to upgrade, otherwise I will look elsewhere very soon. “For Sale: PT HD 2 Accel with G5 dual 2… any offer considered.” Matt Dever d4 Recording Studio ––

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@zackdft 2011. Used under permission.

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ANYONE REMEMBER TAPE?! This print ad from 1996: “Good to know the music will sound just as great ten years after”, not so good to know that you won’t be able to buy BASF tape 10 years after. (Although, it would be remiss of us not to point you to if your stocks of 911 are running perilously low.)

RADIAL CONTROL Vancouver-based Radial Engineering has released its MC3 studio monitor controller, a passive monitor switcher and headphone amp developed as a monitor, sub and headphone management system. Designed for active nearfield monitors, the MC3 is equipped with two stereo outputs and a separate send to feed a subwoofer. Each may be precisely set using a trim control allowing seamless switching between loudspeakers. Separate on/off switches allow any or all to be selected at one time and an adjustable dim switch may be activated to temporarily reduce the monitor level. A single control acts as the master for setting volume. All monitor switching and level controls are passive, thus ensuring no extra amplifiers are inserted in between the source recording system and playback monitors. This is to ensure the signal coming from the recorder to the monitors is not altered in any way. The all-steel-cased switches and steel potentiometers provide maximum lifespan while the discrete circuit topology is supported with a dual-sided mil-spec circuit board and full ground plane to further reduce noise. $349 Amber Technology: 1800 251367 or

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CHRIS JENKINS: BRINGING SSL TO AUSTRALIA Chris Jenkins has been a loyal servant of SSL since the early ’80s. He’s now SSL’s Director of Commercial Applications and recently visited Australia to talk about the new AWS948 board amongst other things. Chris was a ‘graduate’ of the BBC technical departments of the ’70s before joining Townhouse Studio. Townhouse was brand new and Chris was the maintenance engineer. In Studio One there was a traditional custom Helios console and Studio Two was to be the experimental facility with a stone acoustic room and an experimental console from a new UK company called Solid State Logic. The SSL 4000B in Studio Two, as luck would have it, was right next to the maintenance shop, so Chris was on hand to help engineers master this new ‘in line’ style of console which also had a computer and transport control on board. Hugh Padgham, Mick Glossop, Steve Lillywhite, John Leckie, Trevor Horn and others, used the studio to turn out major albums with artists such as Peter Gabriel, XTC, Yes and Public Image Ltd and the word soon spread that this new concept console was revolutionising the way in which music was made. Mix engineers such as Bob Clearmountain, Frank Filipetti, Chris and Tom Lord-Alge now had a tool to develop their careers. After his stint at Townhouse Chris joined SSL commissioning consoles around the world. One of his first jobs was to bring their eight-channel demo console to Australia. This was hooked up to a Lyrec tape machine in a room at the Sebel Townhouse in Kings Cross

and led to the first order for an SSL4000E in Australia for INXS’s Rhinoceros Studio in Sydney. This was for a 48-frame with 24 channels fitted. Of course, the international success that INXS had with Shabooh Shoobah helped the studio gain international recognition and the money to fill the rest of the console and add ‘Total Recall’. Soon after, Chris installed the SSL in Alberts Studio in Sydney and then came the big one for Channel 9 in Sydney. This was the very first SSL 64-channel wraparound console and also the very first SSL with stereo modules, which had been requested by Channel 9 for stereo broadcasting. Their later SSL is the console which ended up being cut down and installed at Sing Sing studios in Melbourne. The requirement for broadcast also meant the development of features such as PFL and fader starts which had never been done before. In the late 1980s the recognition that SSL was not just a ‘flash in the pan’ and had ‘no market in Australia’ (as one of our major importers claimed), also brought the ABC on board. Most of the state capital cities except Perth were populated with SSL consoles. Even Hobart had one for a short period of time. The ABC over the past few years has embarked on an upgrade program and are currently the largest owner of SSLs in Australia. Remarkably there have been very few secondhand SSLs brought into Australia. The most significant one was the 64-channel 4000E brought over to Perth from Onkyo House in Tokyo for Planet Sound Studios when the Japanese company A to Z owned it. This console is still in use at Crank Studios in Northbridge, originally built by the late John Villani. – Cliff Blackburn


New world tour, new drumkit setup


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PRODUCTION STUDIOS FACTS Located at 26 Mitchell Rd, Alexandria NSW 2015

Behind the scenes of AT stories

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Mastering Suite 3

Studios 301 (02) 9698 5888 or

Mastering Suite 1

301 believes the rooms fulfil a niche in the market and recognise that the trend away from big studios will only continue. But there

Of course, sharing the premises of Australia’s biggest name studio has other advantages, with referrals and collaborations part of the life of being plugged into a community, as well as preferential studio rates if you need to avail yourself of one of 301’s mixing or recording spaces.

Mastering Suite 2

Studios 301 has a big idea that’s not so different from the ‘artists’ colony’. They’re in the final stages of building six brand new music production studios at its flagship Alexandria premises in Sydney. Each room’s layout is identical and comprises a 22sqm control room and 6sqm isolation booth. The rooms will be available to tenants 24/7, and are located next to 301’s mastering studios with access through reception. Internet is included, as well as access to the shared kitchen, bathroom and lounge facilities.

will always be a demand for high quality producers, and these producers will benefit from having a professionally designed, acoustically treated space they can bring clients to with confidence. The fact that you can play foosball with other talented producers during the coffee break is simply the glacé cherry.

Mastering Suite 4

If I was a painter (alas, I could never keep within the lines), I’d be well into the quaint idea of joining an artists’ colony. The idea is you do your work in an atmosphere heady with the combined inspiration of other like-minded artistes. You can swap ideas (and partners) while mastering your craft.

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Abbey Road Studios Music Trivia Game!

SHOW US YOUR RACK A while back we asked our AT forum dwellers to ‘show us your rack’ (apologies for the inevitable crassness of the nomenclature… but we couldn’t really call it anything else!). Thanks to Mat from Coloursound, John from Crystal Mastering, Mick Wordley, Brendo, Gigpiglet and the guys from Megaphon for their contributions, and you can see all those on the AT website. But for sheer artyfartiness of the photography, here are our two favourites:

Anthony Manning-Franklin: “From the top you’ve got the Presonus Studio Channel (mod’ed with a different tube), Presonus Firestudio, HHB Radius 30 Tube Compressor, 2 x Moseley TFL-280 and the top of my AKG BX5 spring reverb. The Moseley’s are mod’ed broadcast limiters, they’re über gnarly and snappy, whereas the HHB is the complete opposite and really gentle and smooth – really good for converting changes in level into changes in harmonics/distortion. Then my dbx 266XLs (not pictured) act as the middle ground between the Moseley’s and the Radius 30.”

Dominic McGlinn, DOMC Mastering: (from the top) UA2192 Master Digital Audio Interface, 2 x SSL 502 EQ, custom built SB4000 Shadow Buss Compressor (GSSL) with quad VCAs and low ratios,SPL Vitalizer Stereo MK2-T, custom-made Sontec-style EQ based on the Sontec MEP250EX (‘the rainbow one’), custom designed Crookwood M1-3AI Mastering Console with JLM Audio LED VU/ PPM meter. You can make out a Cranesong Ibis EQ on the other rack, and a custom modified Foote Control P3S Mastering Edition above it.

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Swedish company Pole Position Production is a leading provider of high quality vehicle recordings, providing games developers as well as industry racing simulators with sounds. The company has recorded more than 100 vehicles including jet fighters, tanks, racing cars, old war planes and motorbikes, using a carefully chosen arsenal of DPA microphones. Pole Position used DPAs to record the Lamborghini Murceliago R-SV for Electronic Arts’ Need For Speed: Hot Pursuit (2010). The kit consists of eight DPA 4062 and four DPA 4061s miniature omnis, plus two 4021 compact cardioids. “The low sensitivity 4062s are the big secret for recording exhausts and engines that have a tough sound level,” continues Lohr. “The 4061s do the same job but are used where the sound level is lower. Our secret helper is the BLM6000-B boundary layer mount, which really does the job when it comes to avoiding wind at high speeds. The DPA 4021 has the highest resolution and is used for interior sounds, like inside a cockpit, as long as there is no wind. When recording inside a vehicle, Pole Position usually records eight tracks. “All cars are different, but for the Lamborghini we used three 4062s on the rear wing of the car, two of them close to the exhaust and one a bit further back,” says Lohr. “In the engine bay we mounted both 4061s and 4062s, since you never really know what sound pressure you will have, and there is little time to adjust things when recording race cars.” Pole Position also used its DPA 4062s close to the exhaust while recording the famous

B-17 Sally B bomber for Gaijin’s Birds of Steel (yet to be released), as well as inside the plane by the bomb hatch. “We also mounted two 4061s in the cockpit, the 5100 surround mic in the cockpit, and our two 4021s in the rear of the plane,” explains Lohr. “On the outside we had two 4011 cardioids and the DPA 4017 shotgun. “We also had the pleasure of recording one of a few existing Messerschmitt 109 G-6s recently, with an authentic Daimler Benz engine, and once again the DPAs proved to be outstanding. The recording we got with the DPA 4017 shotgun when this machine flew just a few yards away from us, is totally amazing. First you hear the screaming of the compressor, and then the roar of the engine!” Pole Position started recording racing cars about seven years ago as a fun distraction from music production. “Right from the start our goal was to make the sound as close to reality as possible,” says Lohr. “At that time the games world had its own feel for what a vehicle should sound like, and the sound experience was often not very exciting. Today every high achieving computer game developer knows the sound of vehicles in a game should be as close to reality as possible, and hopefully even a little better. In our recordings we leave our client every possibility to achieve their vision of character and camera view in post production.” Check the AT site for Pole Position recordings. POLE POSITION PRODUCTION AMBER TECHNOLOGY (DPA) 1800 251 367 or

Welcome to the Sweet Spot It’s simply the best way to test-drive PA speakers and studio monitors…

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Find The Sweet Spot for PA Speakers at Soundcorp The Sweet Spot is a purpose-built listening room featuring the best PA speakers from around the world. Inside you’ll find a range of speakers set up ready for comparison. Our staff can help you get started and answer any questions you may have. Come on in and experience The Sweet Spot for yourself, we know you’ll love it.

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Showroom & The Sweet Spot located at 570 City Road South Melbourne 3205 • phone. 03 9694 2600 • fax. 03 9694 2626 • email: AT 19



Virgin ups baggage allowance for musos


LIVING WITH THE WORLD’S SEXIEST SYNTH Every now and then I find myself fixated on a strange piece of gear. It may be an original 1978 Korg VC10 (which took me five years to find) or a vintage microphone, or the Yamaha Tenori-On (that I managed to secure earlier than most after paying a bribe to a man from Texas to ship it to me outside Yamaha’s terms). The recurring theme is that these items are hard to come by. And that’s exactly what got me excited about Teenage Engineering’s OP1 Synthesiser. It was brand new, but that didn’t make it any easier to get. When the Swedish designers from Teenage Engineering released these for sale last year, the limited run of 500 units sold out immediately at the $799 price tag. I recently saw one sell second hand for over $1200 on eBay, so it gives you an idea of the demand.  Here’s a brief overview of what the OP1 packs in its sturdy little aluminium unibody: eight synthesiser modes, a sampler, drum machine, various pattern sequencers/arpeggiator, built-in microphone, G Force motion sensor (like an iPhone) and an FM radio. The unit is charged via USB giving up to 16 hours playing time. USB also allows you to drag and drop your own sounds and samples into it to be tweaked with the odd-ball effects inside. The screen is stunning and offers a range of modes, making programming it surprisingly easy. Using the OP1 is a hoot. And when you think you couldn’t possibly have any more fun, choose the FM radio as a sampler input, dial in a frequency and then set it to Record. It maps out the recording over the span of the keys and you’re instantly making glitchy sampled tunes. All of the sounds and sequences can be recorded onto the ‘Tape Recorder’ which offers realistic analogue type effects, such as slow down, reverse and loop options. The OP1 is an expensive toy. But then, so is my grand piano. It’s a bold and simple unit, yet paradoxically, it is highly detailed. I’ve managed to create blippy eight-bit tracks all the way through to a thick dub-step track for a new Lexus campaign.  Like an acoustic instrument, it demands some time to learn, even though it is represented simply. The OP1 is a world away from the complex, multi-layered patches of a Yamaha or Roland synth, for example, and offers a much simpler building block for making unique sounds. Someone told me recently that for the money they’d buy a MicroKorg. And I agree with them. While thousands of people are all buying a Korg, myself and a few hundred others will continue exploring this strange, quirky and unique sounding machine that operates and sounds unlike anything I’ve used before. – Blair Joscelyne.

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It’s been Korg’s mission to put the thrill of analogue synth performance into the hands of every man, woman and child on God’s green planet. So it should come as no surprise that we now have the new Monotron Duo & Monotron Delay. ‘Go Anywhere Analogue’: That was the theme behind the battery-powered and palm-sized analogue synth that Korg dubbed the Monotron. Now the powerful and fun-to-play Monotron is joined by two new siblings: the Dual Oscillator Monotron Duo offers X-Mod capabilities for generating even more extreme sounds. The Monotron Delay features a Space Delay that can produce intense, analogue-like echo effects. Each is played from a ribbon controller keyboard with newly added features. The Monotron’s simple operation and ease of use (five knobs and one switch) is alive and well, inviting hours of enjoyment and musical exploration. Price: $89.99. Musiclink (03) 9765 6565 or


From $52,100 a pair |

Don’t adjust your sets, the price tags mentioned in this news item aren’t mistakes – just a reflection on the fact that you’re looking at some of the most primo monitoring available. PMC has just turned 20 and is celebrating by weighing in with these two new speakers systems: the stand-alone IB2S-A and the IB2S-XBD-A, which is a two-cabinet configuration with the aforementioned IB2S-A plus an XBD bass speaker underneath. Not to say that for a hefty $52,100 you won’t buy yourself a full-range monitor in the stand-alone IB2S-A. (Perhaps the lads at PMC like a bit of doof-doof in their music?) Regardless, combined, the two speakers could ruin your accountant’s day at $91,200, but you do get 4000W of power, a remote control and plenty of extra low-end extension. Oh, and welcome to PMC’s new corporate blue, but there’s more than just a different paint job. PMC is now manufacturing all the internal electronics of its speakers. Long-term partners, Bryston, deserve a hearty thanks, I’m sure, for its magnificent amps. Neotec Audio (02) 9516 4135 or


The DRS1R/500 is Phoenix’s award winning, class A discrete DRS microphone preamp and DI now made available in the 500 series format, while still keeping all the same sonic characteristics and features as Phoenix’s other DRS microphone preamps. Phoenix come clean admitting that, in terms of design, they had to work hard to find a solution to the problem of powering the units from within the API lunchbox. An onboard power supply effectively increased the API lunchboxes power to a full 24V, which is needed by the DRS microphone preamp. This has also allowed lowering the unit’s current draw to well below normal API requirements and it will work in the vast majority of API lunchboxes including the newest VPRapproved types and other manufacturer’s compatible racks. Mixmasters (08) 8211 6211 or


1 The 24.4.2 provides ten aux mixes; the 16.4.2 give you six. 2 Requires a laptop with wireless, running Virtual StudioLive, and connected to the StudioLive via FireWire. ©2011 PreSonus Audio Electronics, all rights reserved. StudioLive is a trademark of PreSonus Audio Electronics. Studio One is a trademark of PreSonus Software, Ltd. iPad is a registered trademark of Apple, Inc. Freecall 1800 441 440

ith our new free iPad app, “front-of-house” can be anywhere in a venue! Up to ten musicians can control their own monitor mixes1 from the stage. Introducing StudioLive Remote, our free iPad app that works with any PreSonus digital mixer2. It gives you hands-on control of channel levels, mutes, panning, EQ and more for multiple channels at once. Click for a closeup Fat Channel view.

Tweak monitor mix levels, panning and processing. Adjust the graphic EQs, well…graphically. Combine StudioLive Remote with Capture (2-click multitrack recording program), VSL plus Studio One Artist DAW, and you have the most versatile, affordable live mixing and recording solution ever. Get the whole story on our web site. And then get a StudioLive.

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IN BRIEF MASHED UP APPLE ANYONE Steinberg’s LoopMash HD for the Apple iPad is a virtual instrument which made its debut in Cubase and somehow crept its way out of the DAW scene and gone AWOL… or mobile, at least. With LoopMash HD you can blend up to eight four-bar loops together by juggling similar elements across looped beats and tunes. Now ported to the iPad platform, users can choose from over 30 presets and 258 included loops ranging all the way from ethno to electro. There’s high quality content, 19 live performance effects such as tape stop and stutters, intuitive control and 3-D navigation for page browsing. It’s safe to the say that covering Beethoven’s 9th is probably beyond the Loopmash technology, but it’ll be interesting to see what kind of music starts to pop up on the web. Available on the App Store. Musiclink: (03) 9765 6530 or

GARAGEBAND’S ON THE PHONE GarageBand has been released as an app for iPhones and iPod Touch users. In case you’ve been living in a cave for several years, GarageBand features a collection of Touch Instruments that make it easy for novice musicians to play and record keyboards, guitars, drums and basses in a wide variety of styles. Now Smart Instruments allows you to choose from an extensive new library of custom chords so you can play and strum along with your favourite songs. You can plug your electric guitar into iPad, iPhone or iPod touch to play and record through classic amps and stompbox effects, or record your voice or any acoustic sound using the built-in microphone. GarageBand allows you to record and mix up to eight tracks, before you’re forced to keep working on it further in a computer-based copy of GarageBand or Logic Pro.

$FOC |

TonePrints by TC Electronic are like one-shot presets for its TonePrint-enabled range of stomp pedals. So, for example, using the Hall of Fame reverb pedal, instead of mucking around with the settings trying to sound exactly like Bumblefoot (Guns ‘N Roses), you visit the large TonePrint library online and look for a “Hall of Fame reverb TonePrint by Bumblefoot” download it into your pedal via the USB connection and ta-da! Ronny has set it for you, and you didn’t even have to buy him a pint. Get the idea? Okay, now there’s the new TonePrint app. Each TonePrint has been converted to a unique-sounding magnetic impulse that you can beam via an iPhone’s internal speaker, through the pickup of your guitar and directly to your TonePrint pedal. The app contains all TonePrints as mp3 files and you simply place the iPhone’s speaker close to a magnetic guitar pickup when ‘beaming’ and it only takes seconds for the app to do its thing. (Yes, it all sounds a bit Star Trek, but I’m sure TC wouldn’t be making this up.) The TonePrint app is completely free of charge, but users will need the latest firmware version for TonePrint pedals to use the TonePrint App. Currently, you can get the TonePrint App only for iPhone, but an Android version is under development. Amber Technology 1800 251367 or


REAL TO REAL Nomad Factory has launched a ‘truly realistic’ ‘real to real’ [sic] audio tape warming effect plug-in called Magnetic II. The idea is to inject some warmth and character of classic tube circuitry and analogue tape saturation. The plug-in features pure analogue reel-to-reel tape speeds, tape/tube saturation and tape colour effects, as well as a dedicated vintage-style EQ and a built-in Boost mastering section. Intro price: US$79. Nomad Factory:

NEW HOME FOR ABLETON & ANTARES CMI Music & Audio has been appointed the new Australian distributors for Ableton and Antares. CMI: (03) 9315 2244 or

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$99 (Native); $149 (TDM) |

Waves Audio, well known for its professional audio digital signal processing technologies and – it’s worth a mention, by the way – the recipient of a 2011 Technical Grammy Award, has released its H-EQ Hybrid Equalizer. H-EQ is a powerful plug-in for mixing and mastering, featuring vintage and modern EQs inspired by the finest British and American consoles. Included are seven different filter types per band: US Vintage, UK Vintage 1, UK Vintage 2, US Modern, UK Modern, Digital 1 and Digital 2,

in addition to a newly-developed, one-of-a-kind asymmetrical bell filter. An intuitive keyboard graphic lets users choose frequencies by clicking on notes and an exclusive MS Mode allows users to apply different EQ to Mid and Side content. Also featured is a flexible real-time frequency spectrum analyser with multiple display options. Waves H-EQ is available in Native and TDM formats. Sound & Music (03) 9555 8081 or


Fairlight CMI-30A About to Hit


$249 or $999 (Advanced) |

NORD would like to congratulate Manny’s on their 20 years of excellent service to Australian Musicians.

Each module has been updated in the Ozone 5 mastering suite, with refined signal processing algorithms to bring an improved level of mastering quality. The GUI has been redesigned to allow for finer control; the new Meter Bridge introduces a hi-res real-time 3D spectrogram with innovative mix analysis features; BS.1770-2 compatible stereo loudness metering; resizable spectrum analyser and a vectorscope with stereo balance and correlation meters. For mixing and stem mastering, multiple audio streams can be superimposed on a 2D or 3D plot using Ozone 5 Advanced’s Meter Tap plug-ins. Musiclab (07) 3332 8188 or Turrumurra Music (02)9449 8487 or


The Blob... famously a 1958 B-Grade horror where a giant alien blob, resembling most of my mother’s cooking, tries to devour an entire US country town. Notable for being Steve McQueen’s debut leading role and that Burt Bacharach wrote the theme song – which was a hit, for goodness sake – fortunately there wasn’t a serious sequel made. Happily there is to be a sequel of Celemony’s Melodyne Editor, also famous for its blobs, meaning the blob-shaped representation of audio events used in the pitchcorrection software. Melodyne Editor 2 will feature added scale functions, new timing tools and ReWire support. Scale Detective is capable of analysing any sample or piece of music and allows you to transfer the scale and tuning of one recording to another. All very clever, no screaming and running for your life required. Electric Factory (03) 9480 5988 or

161-163 St Georges Road Fitzroy North Tel: (03) 9486 8555 AT 23


5 MINUTES WITH RALF ZULEEG, D&B’S CHIEF ‘FIXER’ AT: There’s no doubt that d&b’s J Series is a great-sounding, brilliantly-engineered PA that deserves its success, but are you still surprised how ‘in fashion’ it’s been? Ralf: I’ll probably make some enemies right now… but, yes, I am a little bit surprised. Especially if you’re looking at the younger guys. I occasionally come across these type of guys at tradeshows: “Oh we just started a business and we’re looking at PAs,” and they immediately go to the Js. “Guys, I’m not sure this one is the right product for you.” A big system is a lot like owning a big fancy car… you feel more important.

Yamaha’s new 01V96i compact digital mixer now offers USB 2.0 connectivity along with new multi-track recording. The 01V96i features 16 in/16 out USB audio streaming at 96kHz, and multi-channel in/out via expansion card or ADAT. The 01V96i shares all the functionality of its predecessor, including 100mm motorised faders, 99 scene memories for instant recall, and fully-configurable userdefined keys. The addition of USB 2.0 enables high-quality multi-track recording and playback, while providing full integration with every major ASIO and Core Audio DAW software package. The 01V96i comes bundled with the latest version of Steinberg’s Cubase AI, offering a comprehensive range of tools for recording, editing and mixing. Yamaha Music Australia: (03) 9693 5272 or

AT: I’d like you to clear something up for me: ‘line source’, rather than ‘line array’, describes a PA that offers a coherent wavefront right into the high frequencies… Ralf: … the idea is to get the coherent coupling up to 10, 12, 20kHz depending on the purpose. AT: Does anybody get that kind of coupling given the wavelength of those frequencies are so miniscule? Ralf: Oh we all do. In theory you can couple up to 50kHz if the array is extremely flat. That’s not the problem, so much. The problem lies more in the arrangement of the loudspeakers, which provide you with an extended Nearfield Effect. AT: Right. You’d better remind me about the nearfield effect… Ralf: The nearfield effect means that for higher frequencies, depending on the distance, you hear just a fraction of the array and the lower the frequency the more of the total array you hear. So to compensate for this you have to increase the energy per box to the very high frequencies – you can be outputting as much as 150dB in the highs in order to get a flat frequency response. Modern line arrays require the latest driver technology, and requires a whole lot of power capability from those speakers. AT: So the HF devices need to be bulletproof. Ralf: The quality of the line array does not rest with the size of the low frequency driver, it’s the efficiency and the quality of the HF unit. AT: Interesting. Ralf: For me if you’re looking at a 4m-long line array, it doesn’t matter if it’s based on a 12-inch or a 15-inch loudspeaker. The age and the package density of the HF unit per metre of array, this is the important factor.

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Digico’s new SD10-24 was primarily designed with the corporate market in mind, offering a smaller footprint console with no reduction in functionality or number of inputs and outputs. There’s also going to be an SD10B-24, a broadcast-specific version for OB trucks where space is often very limited, but a high channel and bus count is required to handle 5.1 audio systems. The SD10-24 measures just under a metre in width, with a work surface constructed from anodised aluminium, overlaid with polycarbonate panels (to resist spilled coffee and potato chip crumbs, no doubt). It has a 15-inch touch

sensitive screen and the control surface features 25 x 100mm touch sensitive faders. The company is also launching the Mini and Nano racks, which offer a wide range of input and output options for any DiGiCo SD audio system. Where the Mini and Nano racks come into their own is that, instead of all the I/O connections having to be in one place, they can be distributed throughout a venue at the most convenient points. Group Technologies (03) 9354 9133 or


Kings of Leon XL8 Lead Vocal Chain


Tiger Woods drops into Oz for a few rounds and suddenly everything’s about golf. Okay, possibly unrelated, HK Audio has released a new portable sound system which it is calling Soundcaddy One. HK has always prided itself on the simplicity of its systems and claims that no other PA can be set up and ready to go in such a short time. HK’s engineers have packed everything you need with this PA into a convenient and elegant caddy (oh… that sort of caddy) and claim that “transporting it from A to B on its built-in wheels is an exercise in speed and convenience”. The mid/high-range unit is mounted and raised pneumatically. Signal sources link into an onboard four-channel mixing console. There is a 600W Class D amplifier under the hood driving 3 x 150mm subwoofers and 6 x 90mm mid/high speakers per side. And you have to admit it looks pretty cool.

Celebrating 20 years of great advice and service to musicians, songwriters and producers nationwide.

CMI Music & Audio (03) 9315 2244 or


Turbosound’s new flagship PA, Flashline, is a complete turnkey system, including line array and sub frequency loudspeakers, control electronics and amplification, rigging hardware, racks, transportation dollies and cases (hey, where’s the gaffa tape?). It incorporates Lab.gruppen’s powered loudspeaker management platform with Lake processing and Dante networking. A new Dendritic/Polyhorn waveguide ensures correct dispersion behaviour in both horizontal and vertical planes, giving uniform audience coverage. The Flashline TFS-900H cab is ‘the first true four-way line array box in the world’, making up a two-box, five-way system with the complementary TFS-900L subwoofer. The TFA-900H delivers 149dB peak from its 11 drive units across four frequency bands, matched to the output of the 20000DP amplifier’s four output channels. The TFS-900H line array module transports and flies fourup and pre-configured from a custom-wheeled dolly. Hills SVL (02) 9647 1411 or

161-163 St Georges Rd North Fitzroy Vic 3068 Ph: 03 9486 8555 The best advice you can give a musician AT 25


SPIKE STENT Mixing Mylo Xyloto Mark ‘Spike’ Stent invites us into his LA studio to talk through the mixing of Coldplay’s latest masterwork. Text: Paul Tingen Photos: Akiko Bharoocha

As so often when a crisis hits, those at the top are the least affected. The digital revolution has largely laid waste to the studio world and the lives of many studio professionals have become much harder as a result. Yet a handful of the world’s top studios are still doing good business, as do a few top professionals that remain relatively unaffected by the crisis around them. Mark ‘Spike’ Stent is one of them.

LA, which features his beloved SSL G Series desk as well as an impressive selection of outboard gear, and his favourite KRK 9000B – “I’m a creature of habit!” he says. However, it turns out that these are not the telltale signs of a champion of analogue gear and/or analogue working methods. Stent is not at all prone to embark on old-fogey style, analogue-is-betterand-digital-is-bad-for-your-health rants.

Stent is Britain’s most successful mix engineer, and currently based in Los Angeles, where he runs his own facility, called Mixsuite LA. It’s a measure of his success that he was involved in the mixing of two albums that topped charts around the world at the end of 2011. He mixed most of Coldplay’s Mylo Xyloto, which went to No.1 in two dozen countries, and the first two singles, Shake It Out and No Light, No Light, from Florence & The Machine’s Ceremonial, another album that was one of the major events of the year. In addition, in recent years Stent had his hand in hit albums by Duran Duran, Keane, Christine Aguilera, Usher, Lady Gaga, Muse, Massive Attack, Cheryl Cole, Madonna, Beyoncé, Björk, and many more. It’s an impressive track record that, while heavy on UK rock bands, also sees him crossing over to American R&B/hip-hop territory – a rare achievement for a UK mixer.

Instead, the Briton stresses, “You have to move with the times. I still use my analogue gear, and particularly my magic SSL desk, which has old E-series cards in the centre section and a sound that I love. I like the sound of analogue in general, and the items of outboard here, but I have gradually moved to doing more and more things in the box. It has been a natural progression, in part driven by the fact that people now call me days or weeks after I’ve done a mix, asking me to do a different version. In five minutes it is on its way to them. In the old days it would have taken three hours to recall a mix on the outboard and the desk. So I have embraced new technology, while not forgetting my analogue roots.


Via an echo-y phone line and then a dropoutladen Skype connection (21C communication devices also suffer from the ills of digital, it appears), Stent elaborated on his working methods on general, and those on Mylo Xyloto in particular, in the process offering some pointers on how to thrive in the digital age. He was talking from his control room at Mixsuite

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I work with a hybrid of old and new approaches. It depends on the music, and I’ll sometimes do urban tracks entirely in the box. But colouring is very important to me, and when you’re working purely in the box it can sound too clean quite easily. Anything more organic, more rock-like or band-like, shouldn’t sound too clean or nice, so it will often benefit from being put through the analogue domain. But it’s very rare these days that I lay an entire session out on the desk. Instead I’ll start in the box, and will then lay out sub-mixes of selected tracks over the desk, most often drums and bass. It gives me a particular colour that I can’t get any other way.”

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I have gradually moved to doing more and more things in the box… people now call me days or weeks after I’ve done a mix, asking me to do a different version

FROM TEABOY TO SO-CAL Mark ‘Spike’ Stent’s roots go back to the 1980s, when he got into the studio world via the then regular tea-boy-to-engineer route at Jacobs Studio in Hampshire, just outside London. He earned his nickname because of his spikey hair at the time, and one rock ‘n roller, unable to remember the name ‘Mark,’ called him Spike. The spikey hair has long gone, but the nickname has stuck. Stent went freelance in 1987 and made his breakthrough with pioneering mixes of the KLF. Since then he’s clocked up an astonishing 800 credits and counting, and earned three Grammy Awards. He worked at Olympic Studios from 1999 to 2007, after which he built a studio in Salisbury in west England, which, he says, was gear-wise a ‘duplicate’ of his setup at Olympic. However, after a sixweek stint in Los Angeles in the summer of 2007, Stent and his large family (wife and four children) decided that they wanted to stay there, which meant that he gradually moved his operations over to LA. He first worked from Chalice Recording Studios in LA but brought over his beloved SSL G-series in August 2010, when he began work in his current Santa Monica facility, which consists of a control room, a main live room, a small booth and a machine room. He works there with assistant Pierre Eiras and former assistant and now fullyfledged engineer Matt Green.

UNDERSTANDING BIG BOTTOM With very high profile mixes, like those of Coldplay and Florence & The Machine, under his belt, Stent is clearly going from strength to strength, despite the challenges that moving to LA first entailed. “Moving to American was a big change for me,” he says. “I went from being a bigger fish in a smaller pond, to a minnow in shark-infested waters over here! I had done a lot of high profile work in the UK, and some people knew me here, but I had a point to prove in America. It gave me a big kick up the bottom, so to speak, which I think was good. I was in the habit of doing things a certain way in the UK, and needed a challenge. Coming to the US was a watershed moment for me. I’d been coming and going here for 20 years, but actually living here and properly absorbing everything was different. I had to learn very quickly how to mix urban records, and what was surprising to me was how different the approach to bottom end is here compared to the English approach. The way they place and EQ the kick and the rest of the drums, and the vocals… everything is different. Of course, I’d listened to American records for decades, and I thought I understood the sonics, but it turned out I didn’t. There’s far greater depth and punchiness to the bottom end in American records, particularly on urban records and in general they like things a lot darker. The kick is mixed much further forwards, and the snare and/or clap further back. Coming here made me understand the sonics of American records, and I had to adapt.”


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“I wanted the drums in this song to feel ‘hip-hop’ rather than ‘rock-y’ and ‘room-y’, so I added some kick samples. At the same time you don’t want it to be too ‘sample-y’, otherwise it won’t sound like a band thing. On hip-hop and modern records in general there’s a toughness to the sound of the beat, and I wanted to have that in this song. I sent my idea to the band, and they liked the approach, so I continued with it. The toughness of hip-hop is mostly about the sound of the bass drum, but there’s not one particular frequency that needs emphasising… it depends on the track. I had a variety of plug-in compressors and EQs on the drums, among them the SSL-Channel, the Waves VEQ4, the UAD Fatso, and the CLA-76. So I got a mean vibe going with the samples and the plug-ins and I then laid the drums out over 12 channels on the desk, with different sounds on each channel, and I bussed stuff to outboard Neve EQs, Distressors, maybe the Fatsos, and then back into the console.

One of the secrets of Stent’s success appears to be that he’s able to adapt well to changes in environment as well as in technology. His current hybrid method of working – integrating the old and the new – means that he has “shitloads of plug-ins” and in general has made himself completely adept at working in the box, something many old-school engineers baulk at. “I haven’t bought anything new on the outboard front for a while – the Bricasti was the last piece of kit I bought, nine months ago,” explains Stent. “If you keep changing your gear, it’s hard to focus on the music! But I am always getting new plug-ins. I like the UAD stuff and the Waves bundles, and particularly use the R-Bass and the JJ Puig and Chris Lord-Alge plug-ins. SoundToys make some really great things – I use their EchoBoy and Devil-Loc plug-ins a lot. There are many great things going on in the box. I don’t have a controller and still use a keyboard and mouse. I’ve become used to it. I’ve had a ProTools system for many years, so it’s not a new thing for me to work in the box. It’s been a natural progression, just like ProTools has gradually become better, going 24-bit and with higher sample rates, computers being more power and memories being larger. It means that everything is so much better than it was in the past.”


Stent’s successful and apparently seamless adaptation of new working methods is a central theme when he describes his work on Coldplay’s Mylo Xyloto. The album was produced by Markus Dravs, Daniel Green, and Rik Simpson and recorded at Coldplay’s The Bakery and The Beehive’s studios in North London – the latter is in a big hall, offering ample possibilities for big room sounds. Stent had a hand in mixing 11 tracks off the album. When asked for details, Stent describes a very fluid way of working in which all aspects of the production process run into each other. Tracks didn’t evolve in a linear way, and the mixing stage wasn’t necessarily at the end. It’s a modern approach that he, again, is entirely comfortable with. “The band was still recording when I was mixing, so versions would be changed and updated as I was going along. There were also different mixes, and that’s the way the band likes to work. I was not privy to whether they edited different mixes together or not. “I’d say that 90% of the mixes were done at Mixsuite LA, and Chris would redo a vocal part while I was working, or Johnny would change a guitar part, and I’d work those changes in as I mixed, or after I’d finished the mix, and I’d send that back to them, and they might react to that, and change things again.

ROOM SOUND OPTIONS “One of the main things with this [Paradise] and many other tracks on the album was that [producers] Rik, Dan and Markus had recorded the room sounds in an amazing way. There were many room sound tracks, so I was lucky in that I had a lot of choice in how I wanted the drums to sound. There were Royer and Coles ribbon mics and other room sound tracks, and when I stuck them in a compressor they would open up quite large. I did a lot to colour the various drum sounds. I balanced all the room sounds in the box, and they would then come up on two channels on the desk. My main aim in the analogue domain was to try to get the bottom end big and punchy and powerful, and make sure that the room sounds were warm, rather than harsh and small. “After I worked on the drums, I brought in the bass, which was split over four channels on the desk: synth, real, sub, and DI. There were three pairs of guitars and a lot of the processing on them happened in the box. I had the Waves SSL Channel and PuigTec for EQ on the guitars, as well as the PuigChild 670 and CLA-3A compressors, and a SoundToys EchoBoy delay. There’s a lot of compression going on. I like compression. I was just trying things out, trying to get some more colours into the guitar sounds and some delays. The guitars then went through the desk with more colouring from EQ and compression. There were some keyboard and string tracks, each of which came up on the board as a stereo pair. I tried to keep those elements simple. The vocals were treated quite extensively in the box, with the Waves De-Esser, SSL Channel and C6 EQ, and PuigChild 660 compression. I then sent the vocals out to the desk, in which I had three hardware insert chains, each with a dbx 902 De-Esser, a Distressor and either a UREI 1176 Blue Stripe or an LA2A, or sometimes another compressor, like a Fatso.”

VOCAL CHAIN For Chris Martin’s lead vocal Spike uses a combination of a Waves De-Esser, Waves PuigTec EQP-1A, Waves PuigChild 660, and the SSL-Channel. “I have different EQ in different sections of the vocals, and automated de-essing. If you’re really compressing the vocals hard, things like breaths need to be volume automated in ProTools, so you’re not hearing the breaths”

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Spike’s ‘magic’ SSL G series mixing console, which has the older E-series cards in the centre section and “a sound that I love”. You’ll spot Spike’s trusty KRK 9000 monitors on the meterbridge along with a pair Yamaha NS10s, and the more esoteric Barefoot Sound MicroMain 27 monitors.

G SERIES: SOUND OF PARADISE “With Paradise, almost everything went through the desk. With all songs I sent the stereo mix through an SSL Quad Compressor and Massenburg 8200 EQ, and then via the Apogee Symphony converter back into the same ProTools session, which was at 24-bit/48k. Later on I also did a stems mix to a separate ProTools rig at 24-bit/88.2k. After the project has been mastered I usually do a stems mix, because some people use them for playing live, and they can be needed for different versions or remixes. I don’t put any limiting on the stereo mix, although I do send out heavily limited files, using the Waves L2, to the band, producers and record company for feedback. They need to be able to compare my mixes to stuff that they hear on the radio. Quite often I get asked to send limited and unlimited provisional mixes. But mastering always gets the unlimited files. I don’t normally become involved in the mastering process, unless the artist specifically asks me. But I have a high turnover of work, and by the time they’re at the mastering stage I am already busy with the next project.”

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“When I mix an album, I look at mixes on a song-by-song basis, I never look at the whole album. I don’t make templates. That doesn’t work for me. In each song the band may track things differently, or use different mics, or play different instruments. I may sometimes try to retain some settings that I really like, but I never actually use them. I go by feel. I know what I like and how I think it should sound. I may have compared the bottom and top end of some different mixes, when I’m concerned about something, but I always just tend to go with what I think is right for each track. I don’t worry about consistency over a whole album. That’s the job of the producers and mastering engineer.” FEELING IT

“Because feel is so important, I always ask for the rough, and also for the very first demo a band or an artist made, and any other version that the band or the producer, or the A&R man thinks is relevant. I may end up with two or three versions as reference points, which I’ll then load into the session, so I can flip between them and compare them. The original demo is there for the vibe – when people spend a lot of time on production and overdubbing they may lose sight of that. You can sit there and be a purist and say that the snare drum sound isn’t right, but you have to listen to the song as a whole. This obviously wasn’t an issue with Coldplay, but these days I get stuff

that’s been recorded in bedrooms, in small programming rooms, on stage, and that may have huge problems that need sorting out. If that is the case, I’ll dive into the playlist, and try to get something together that I feel is a better representation of the track. If necessary, I’ll program new parts in, or I’ll bring someone in to program them. I have never sent a mix back because I felt it wasn’t ready or not good enough. If the song isn’t working, it’s my job to sort it out and get it right!” CREATURE OF HABIT

Clearly, Stent’s flexibility and ability to adjust to the needs of the times, and the song he is mixing, stand him in good stead. It’s interesting therefore, to hear him calling himself “a creature of habit” when referring to some of his gear, like his KRK 9000 monitors, and also to the way he likes to lay out his sessions. When he gets a ProTools session in for mixing the first thing that happens has remained relatively the same for a number of years, which is that his assistant will begin with, insofar as required, “cleaning the session up and organising the whole thing in the way I want it organised. I am quite anal about the way I like the session organised, because it makes the whole process quicker and easier. Vocals are cleaned, breaths taken down, and any de-essing is taken care of as well, using the Waves de-esser. If I want to compress the hell out of a vocal because of a certain vibe I’m after, I need all the breaths


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Hybrid Approach: Spike will do a lot of his mixing ‘in the box’, combined with hardware processing and the sound of his SSL mixing console. (Below) One of the juiciest racks going includes: three Distressors and a Fatso, six SSL E Signature Channels, a GML 8200 parametric EQ, The Phoenix valve compressor, a Culture Vulture, and a TG12345 Curve Bender EQ from Chandler Ltd.

CONSOLE BREAKOUT “After getting an initial balance in the box, a lot of the session was broken out over the console, with these internal submixes being bussed to various different channels. Like there’ll be a kick channel, and a snare channel, toms, overhead and room channels and any other spot mics, on which I will have had things like Neve 1066 EQs and the Thermionic Culture Phoenix compressor. I’d also have three separate drums groups on the desk with Distressors and outboard [Empirical Labs] Fatso and SSL compressors, which would then be mixed in with the submixed kick, and snare, and tom and room channels that are coming up on individual channels on the desk. So I’m getting a mixture of these individual channel sounds and the really compressed and EQ’ed group sounds. It was a similar process with other instruments, with a submix coming up on an individual channel on the console and then a separate effects group track being mixed in with that.”

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taken down, the ends of lines cleaned up, and if it was recorded in a large room and/or with a handheld mic, all the extraneous noises removed. After everything is cleaned up and organised, I’ll add samples, if needed, usually kick, snare and tom samples. I never replace, I always add to things.” “When I begin the actual mix, I’ll usually start with the drums. I’m of the opinion that unless you have a solid foundation, it doesn’t matter what you stick on top of it, it won’t sound good. So that’s where I start, depending on the type of song. Once I’m happy with the drums, I’ll bring in the bass, then the guitars and keyboards, and eventually the strings and so on, again depending on the nature of the track of course. During this process I’ll periodically check that the vocals sound okay in the track. Once I have the instrumentation as exciting as I can get it, I’ll bring the vocals in, and I’ll really fine-tune them. Throughout this process I focus on the general vibe; I’m not very clinical in my approach – I want to make sure that I get the emotion right. Matt will tell you that there have been times when I’ve pulled everything down three or four times and started from scratch again each time. I do this, rather than tinker with what I have, when it isn’t working, when the vibe isn’t right. All this tends to happen in the box, and, as I described earlier, I’ll then lay different things out over the desk, to get that analogue colour.” “When you look at the screen shot for Every Teardrop Is A Waterfall you’ll notice the demos/roughs at the top, for reference. I did indeed begin the mix with the drums, as I always do, but then brought in the piano, because the piano provided the most important hook in the arrangement. After that I added in the bass, then the guitars and then the whole end section where the real kick drum comes in. I didn’t add any samples, this track was mostly

a case of balancing what I had. The main thing in the song was the piano riff, and the main issue was keeping the warmth and excitement going all the way until the end. There were some strings, but they were easy, very straightforward to mix. All the treatments that you hear were already in the track, they were part of the production. The production was really clever. The whole orchestral feel, the fact that the arrangement sounds quite big, came from the producers, ie. Markus, Ric and Dan.” “When working in the box, I’ll tend to subgroup the different kick, snare, toms, and other tracks. So any kicks will be subgrouped to a kick master, the same with a snare or clap, and all that is then sent to a drum master, on which sub compression is happening. So there is a lot of submixing and processing, which mostly is about compression. The plug-ins I used on the drums in Teardrops were mostly Waves SSL Channels for EQ and UAD Fatsos for some colouration. I probably also used the Fatso or Devil-Loc plug-ins for colouration on the room sounds, just to give some warmth and distortion. I had quite a number of plug-ins on the piano: the Waves SSL E-channel and SSL G-channel for EQ, the Waves Puigchild 660 for compression, and the SoundToys FilterFreak, Decapitator, and EchoBoy. The Decapitator added some drive to the sound, the FilterFreak changed the nature of the tone, and the EchoBoy added a delay. A lot of the sound is coming from the FilterFreak and Decapitator. On the bass I had, again, the Waves SSL channel for EQ, and the CLA-2A compressor, plus the SPL TwinTube, a plug-in that I really like. The lead vocals had a Waves De-Esser, Waves PuigTec EQP-1A, Waves PuigChild 660, and once again the SSL-Channel. I have different EQ in different sections of the vocals, and automated de-essing.”



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Caleb, Matt, Jared and Nathan Followill are back in town. Text: Christopher Holder Show Photos: Marty Philbey

The British music press can’t resist a good story. When it comes to bands and musos, sure, being able to hold a tune would be handy, but a colourful back story floats more boats than virtuosity. So when Kings of Leon emerged, like the remnant of a lost hillbilly tribe out of the Smokies, the UK press were fairly falling over themselves to find out who these boys were. With their long hair and beards, were they the result of a mishap at some secret Clearance Clearwater fertility clinic? Were they really brothers? Were they sons of a preacherman? Had they duelled with the fiddle-playing devil and won…? Turns out the back story was a far more mundane and familiar one: endless gigging in clubs in and around Tennessee. Slowly building up a following. The van they gigged and slept in found itself pulling a trailer, and then the van was traded in for a bus pulling a trailer. Kings of Leon has been criss-crossing the world ever since. And front of house engineer, Brent Rawlings, has been there since the very beginning. Brent Rawlings: 10 years ago it was just myself, ‘Nachos’ [Followill, a cousin of the brothers] and the band, driving a van between Nashville, Atlanta and Birmingham. We did that for a while and it just started to build from there. That period helped the band find its groove before they broke in the UK. They sent a song over and they had a hit on UK radio with Molly’s Chamber. That was our first taste of something bigger. AT 34


Brent, like any engineer of a hard-working band on the club circuit, made the sound work with whatever he encountered. But when success started to come the band’s way he had the luxury of hauling his own kit. BR: I didn’t carry for a long time. But I will say that from very early on I noticed that anytime I plugged a mic into a Midas console it instantly sounded good. So for a while there I would carry just a Midas desk for a club tour and otherwise use what they had installed. And then we’d carry the monitor speakers to maintain consistency for the band. And then we got the band into ears some time later. CH: So the guys weren’t obsessed with being the loudest band in the world? BR: A long time ago we talked about limiting the stage volume. Sometimes a band comes off stage at 120dB, which doesn’t do any good for the guy out front being asked to stay at 110dB. Just take a look at Jared’s bass rig. He uses two 8x10 bass cabinets – that’s a beast. CH: So you confronted that fairly early on in the piece? What can we do to keep stage sound under control? BR: Yeah. I remember a couple of times I would turn off the entire PA and all the monitors and explain to them that we should try and make it sound at least decent like this – with the instruments only. With the PA and wedges off, the drummer is the loudest sound on stage because he’s beating something with sticks. Now just turn your amps up enough to hear yourself over the drums. And then we’ll

AT 35

start ‘reinforcing’ the sound from there, instead of automatically plugging everything in and turning it all up to 11. Do it that way and your perspective gets skewed. CH: And there’s a flow-on benefit all the way to the FOH PA and the audience… BR: I mix at around 102 or 105dB A-weighted – real comfortable. I never have a problem with the sound police. I use a lot of compression and can make the maximum volume sit right there, if necessary. TUNING THE PA

CH: Power and impact is important in your mix but you’re being especially careful not to let rogue volume spikes skew your mix? BR: Right. I’m constantly monitoring Smaart [real-time analyser] to make sure nothing’s crazy out of line. CH: And how do you like to tune the PA? BR: After our system tech is happy with the sound of the PA in the room, I tune the PA with the vocal mic. That microphone is the hottest instrument, so I bring it to the FOH position and tune the PA to that and get it sounding as sweet as I can. Which is pretty old school but, logically, if that’s going to be the loudest mic in the room… And Caleb is a hard one. It’s not like his singing is always coming from the guts. Sometimes it’s coming from the throat; sometimes from the nose. He’s got a lot of middle and a lot of high-middle and sometimes it’s pretty honky, so making it loud above a rock band and still having a rock show that pleases an audience – that’s quite a challenge. VOCAL CHAIN

CH: Can you talk me through Caleb’s vocal chain and how you get his vocal to sit high in the mix? BR: Right. So, like I said, I tune the PA to that mic, which is the first big step. I then do my best to make his vocal as ‘big’ as I can. I make a big cut in the middle, usually somewhere between 800Hz and 1.5kHz. Then I run it into the XL8’s three-band multi-band compressor, as a channel insert. I let the low section be wide open and the high be wide open, I’ll dial in the middle frequencies and squash it real hard. Somewhere between 600Hz and 1.5kHz. Next is the channel strip’s compressor which I’m only working a little bit – the only one that’s working really hard is the mid-band compressor because he’s just got a lot in that area. CH: How are you creating space for that vocal sound in the mix? BR: By panning the guitars really hard. So hard, in fact, that the side hangs of the PA are ‘opposite’ – you have a main left and right arrays but the sides are the opposite image. CH: Sure, so people out to the side are still getting a stereo image… BR: And then on the floor are some small fill boxes and they alternate as well – left, right, left etc. I pan the guitars to the extreme left and right AT 36

unless Matt’s doing a guitar solo, then I’ll put it in the centre and bring him up. So basically my strategy is just using stereo to my advantage and making the vocal centre. Kick, snare, hat, vocal – that’s my centre. CH: Do you carve out space in the mix with EQ to give the vocal more room? BR: Yeah, definitely. I carve out a big chunk of the low-mids of the bass guitar – especially because the low-mid in these arenas is so muddy with room reverb. Same with the kick drum. In fact, I go through and cut the mid frequencies on the instruments as much as I can get away with, while still sounding musical, then it usually leaves me with enough of a hole for the vocal to hold its own. RIDING HIGH

CH: With the vocal mic so hot, do you leave the vocal mic on the whole time to preserve the overall tone, or duck it as much as you can? BR: No, I’m definitely riding the vocal fader. I’m using the XL8 desk scene automation to recall effects and mute groups – delay times will change every song and the mutes will come on and off depending on what’s getting used. From there, I’m definitely playing the VCAs a lot and pulling the vocal mic down when Caleb’s not singing. But you’re right, it’s the hottest mic and it’s getting a lot of stage sound – it’s like your ‘downstage overhead’. So I won’t duck it totally, otherwise the overall sound of the mix will change too dramatically. CH: You just pull it down as much as you dare? BR: There is so much hat and snare in that vocal

BASS SOUND Brent Rawlings: With those 16 x 10-inch drivers in the bass cabs you’re almost guaranteed that a certain frequency in the room will build up. The last thing in the line before Jared’s bass head is an Avalon 737 which we’re using to take a very narrow frequency out of the bass. We find it daily – just listen and think, “no matter what he plays, 113Hz is ridiculous – so we’ll notch it hard.” And it helps for the bass player as well because then his tone is a lot cleaner.

Caleb’s Vox AC30s and Matthew’s Ampeg Reverberrocket combos, both miked up with a combination of a Sennheiser e906 and Audio-Technica AT4050.. Front of house engineer Brent Rawlings with his Midas XL8 digital console.

mic, and there’s nothing I can do about it. When he steps in front of the vocal mic it’s better but as soon as he moves out of it, then that fader’s coming down a few numbers. CH: What else are your fingers hovering over? BR: The XL8 has a bank of user-assignable controls and I’ll assign, say, the lead guitar pan to one pot, because I’m constantly putting Matt into the centre for a solo and then pulling him off to the side when Caleb’s on the mic. And then I have one assignable pot set to be the high-pass filter of Caleb’s guitar channel, because sometimes his tone will get a bit muddy and I’ll need to take a bit of the bottom out to help it cut through in the mix. Other times he’s barely playing it and I’ll need that bottom to come back in. So it’s just stuff I’m having to grab all the time, and it’s great to have that sitting close, just to the right of the VCAs. CH: Sounds like you’re really making the XL8 your own. Are there times – such as at festivals – when you’re forced to mix on other consoles? BR: Well, fortunately, the minimum we’ll carry is Control, meaning: FOH desk, monitor desk and monitor wedges, our own microphone package, and our own backline. We’ve got it all multipinned and ready to roll. So we can be out, up, patched and checking in 10 minutes. And when I say ‘we’, I’m sitting up at the front of house position, nervous, waiting for the guys onstage to get it done. But the team up there is awesome. So then Saul [Kouta, on monitors] and I just get a line check and that makes it easy.


Saul Skoutarides is a young, switched-on Aussie engineer who flew the local coop three and half years ago – after a stint at Johnston Audio – to try his luck in the Old Dart. He scored a gig with Britannia Row, taking on the job as a tech for a Kings of Leon tour. He’s been kept on ever since, eventually taking over the monitor reins in early 2009. His main weapon is a Midas XL8 mothership. On stage he relies on Future Sonics in-ears, Sennheiser wireless, d&b M2 monitors and Q Subs. Saul Skoutarides: I don’t overly program things. I use snapshot automation for mutes mainly. I tend to work in a more analogue fashion, which is why I like this board, you can work in that way. I’ve got guys on wedges, ears and both, and I use the desk to make life as easy for me as I can. CH: I gather Caleb opts for only one ‘ear’? SK: [With a tone of resignation] That’s right, he’s half on ears. I inherited the gig that way and there’s no point trying to win Caleb around at this point. The lead singer of a band he respects got it into his head that one ear is always better. All I can do is the best job I can in the situation. CH: For those who don’t know, using one instead of two earpieces makes it harder to give the muso a good mix and they tend to have to crank the level of that one ear. Is that right? SK: One ear is not a good idea – you can damage your hearing because you have to run it so loud. But I’ve found that I’ve not had to run this particular system blaringly loud. I think that’s because the Future Sonics design is ported (you

DRUM SOUND Brent Rawlings: I base everything off the Rode NT4 stereo mic I use as an overhead. It’s not a mic with an amazing flavour, it’s just that it’s stereo and its phase coherence is perfect, because of the fixed X/Y capsule configuration. Therefore, if I put it right above the drummer’s head it’s very true to what he’s hearing and provides a very realistic picture – the snare sound, cymbals, everything sounds very true and provides a good overall mix in itself. So I listen and start everyday with just the Rode NT4 and the kick drum, and I’ll get those sounding as good as I can. And then for some better treble on the cymbals I’ll mix in some of the AKG C414s that are positioned closer to the cymbals. The Sennheiser e904s on the toms, and on the snare we have a Shure KSM32 on the top and a SM57 on the bottom. And here’s where a digital desk like the XL8 really comes into its own: I treat the Rode NT4 overhead as ‘zero time’, delaying the other mics back to that mic, so that the snare mic and overhead mic are firing at the same time. So, for example, I’ll have the overhead mic cranked, turn the snare mic on and then grab the delay time of the snare, adjusting and listening for the sweet spot. That works really good for me. It’s helped me with my snare sound tremendously and the toms. I can get them up in the mix a lot easier but it’s not eating the mix alive.

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JPJ J SERIES PA JPJ’s Craig Gordon talks us through the rig: It’s a d&b J series PA. Eighth Day Sound is the principal provider (from the US) and they have a system tech for FOH and monitors on tour. They rent ‘stacks ’n’ racks’ from us. The main hang is 32 Js in the main and 24 in the side hangs. There are six J Subs aside in the air and nine B2 double 18-inch groundstacked subs aside.

The J Series line array really comes into its own when you fly the J Subs. It takes some of the hard work out of the main array (helping to clean up the overall sound), provides pattern control and of course provides some real low-end extension. The B2s are out and out subs. The Rod Laver Arena and the J System work well together – you can get a really tight low end.

can adjust the port to augment/attenuate the bass response), so it’s not totally sealed in your ear. I think that helps. CH: Are you sending the same mix to his wedges and ears? SK: No, I don’t have everything in both. I have the drums in the wedges. I have the vocals and his guitar in both. The other instruments are in the wedges. CH: Have you also inherited the mic selections? SK: We regularly reassess our microphones. For example, our vocal mic used to be the Sennheiser 935. Caleb has a mid-heavy voice – the 1k to 3kHz range can be quite full-on in his voice – so we switched to the Sennheiser MD431 dynamic mic instead. We quite like the 431. It’s harder to come by and, although Sennheiser has been good to us, we’ve ended up having to buy most of the 431s because Caleb has a habit of breaking them. Not in anger, but simply as part of the show – one will take a dive at the end of the night. The construction doesn’t lend itself to handling that sort of treatment. I’ll pick up the mic at the end of the show and hear a tell-tale rattle. Otherwise we use a mixture of Shure, Sennheiser and a smattering of Audio-Technica. SOMETHING FOR EVERYBODY

CH: Can you elaborate on your analogue-style mixing technique on the XL8? SK: I’ve laid out the board such that I have a load of submixes going on. This approach gives me more flexibility. I then use the matrix section as my masters. I can do that because on this console the matrices are almost exactly the same as auxiliaries, although you can’t assign them to the AT 38

The d&b D12 amps have processing and limiting onboard. There are some useful settings in there, such as a coupling filter for when you have a big hang, or a high boost for when you’re throwing further than 80m. We then have two Dolby Lake Processors for time alignment and routing.

L/R bus. So I’ve got a bunch of submixes that go into the matrix masters, which means I can have a lot of things on the surface without having to go hunting for them.

quickly. Mainly they’ve been cable issues. If anything internal goes wrong it’ll usually happen in the truck and you’ll notice it on start up – once it’s up and happy, it stays that way.

CH: Apart from the main ‘culprits’ being right in front of you, how does this approach provide you with more flexibility?

One thing you need to keep an eye on are the Ethercon connectors. The AES50 audio protocol that links all the units runs on Ethercon. Pins 1 and 10 give the sync. If something gets bumped, and it loses one of those pins for a second, you can lose sync. The answer is to unplug the cable and plug it back in again.

SK: Okay, for example, Matrix 1&2 are my main stereo mix for Caleb’s ears. Aux 1&2 are a stereo guitar sub mix. They bus into Matrix 1&2 as a submix of his guitar only into his ears. That means I can EQ his guitar in his ears only. I’m not performing the EQ on his guitar input channel, which is going to affect everyone else, only his ear, via that aux. The same applies to his wedge mix. His guitar EQ is just his guitar in his wedge only. CH: Sounds like the XL8 has really won you over? SK: It’s the sound, the user experience, and the thought that’s gone into the surface that really impresses. For example, the pots are actually analogue with an A/D converter under the surface – it’s a combination of the pot and the software. It means that if you do something really slowly it happens really slowly, as opposed to other systems – when you slow down with a digital encoder nothing happens because the steps between the info are a set distance. Or when you’re sweeping a frequency in the EQ they’ve emulated the whole phase shift – it sounds like an analogue EQ. Which is probably why the board is so expensive, the hardware is uniquely… complicated. Saying that, I’ve only had it fall over a couple of times, and in ways that could be sorted very


Kings of Leon is a class act and the d&b J Series array was sounding great in the Rod Laver Arena. Sure enough, seeing Brent in action, he’s a busy man on the VCAs. Maintaining a strong vocal presence is of paramount importance to him and it shows. A big powerful drum sound was clearly evident, although not overpowering. I think Rode will find itself selling a few more NT4s as a result – Brent’s technique (although he’s by no means claiming it as his own invention) is well worth trying to emulate [see the Drum Sound box item]. There’s plenty of power and control in the drum sound. The word on the street is, Kings of Leon will be shortly having a long layoff after 10 years of constant touring. Richly deserved… I’ll have to find out how Brent will cope. I hope he enjoys fishing. www

Take a look at the AudioTechnology YouTube channel to watch Brent Rawling talk us through the Kings of Leon vocal sound.



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Nick Arundel, Rocksteady Studio’s Audio Director and Composer, adds a bunch of new tools to his utility belt for the latest Batman outing. Text: John Broomhall

Videogame audio productions don’t come much bigger than Batman: Arkham City. With over two hours of orchestral score, thousands of sound effects, and 13,000 lines of dialogue, populating some 40+ hours of game-play, it is nothing short of epic. Acclaimed development studio Rocksteady is now shipping its second Batman title. Riding high on the retail success and critical acclaim of the first game, Nick Arundel, Audio Director and Composer, who has been with Rocksteady since its formation in 2006, entered the second marathon with some clear objectives for raising the audio bar, as he explains: “When you finish a big project, you often just think about the ‘ones that got away’. We’re not the kind of team to rest on our laurels, thinking, ‘oh yeah, we did a really good job’. We’re thinking about what we can do better next time. We’re always pushing ourselves to be as good as we possibly can be. One area I wanted to work on was the long introductory sequence at the outset of the game. Previously, the music felt like five lots of three minutes whereas this time I wanted it to feel much more like one continuous piece (even though there are loops and holding points if the player stands still), so we created a much more focused experience. “I also felt we could be more extreme with the dialogue processing – something we’ve really ramped up this time – not just mastering, but post-processing by hand to give certain effects to certain characters. On the previous game, we had custom batch processes for characters, but this time around, we’ve hand-rolled it per line with specific effects

on individual words. For instance with The Riddler, a key word in a sentence has been processed more than the others around it – highlighting it. For example, whenever he says ‘Batman’, he kind of stutters it – but it’s like an electronic stutter created by cutting and pasting x number of samples with their own reverb and delay added, which trails off over the rest of the word, ‘Batman’.” GUN REPORTS: FIRED UP

One of the most significant shifts of focus for the new title was away from library source to bespoke recorded sounds. Previously some 70% was library source; this time around, there was less than 5%. The increased scope of location and foley recording took team members to some diverse locations including a full-scale steel mill in the north of England, the White Scar caves in Yorkshire, and Dingles Heritage Centre, a fairground museum in deepest Devon. Meanwhile, in the US, Andrew Riley and Bryan Watkins made extensive ambience recordings in the desert and in downtown Los Angeles with Watkins also conducting a comprehensive gun-recording shoot on Warner Bros.’ own film lot. Though gun sounds are not a primary focus – with Batman never personally firing a gun – they nevertheless become very important when aimed at him during gameplay, so Arundel set about some new audio capture: “I was really impressed with what Bryan (Watkins) had done with the shootout in the woods in Public Enemies. I thought the dramatic perspective changes could really work in Arkham – so we decided to do our own gun session on the Franklin Street set. We had nine mics in the middle of AT 41

a long street running down to a T-junction 200 yards away, where there was a further recording setup around the corner. Meanwhile, to the right of the main close-mic cluster, we had mics up in the ‘buildings’ to get interior reflections... We also put a surround recording rig in a courtyard another street away – the idea was to get dramatically different perspectives.” ON THE MARK

The result of a long, sunny day’s firing of an impressive array of gun hardware in Hollywood was a huge ProTools session with everything synced up and mix perspectives carefully laid out. The close gun recording was used for a range of 10 metres and then two further perspectives were chosen from the total of five available. These perspective choices then figure in gameplay helping the player readily discern enemy positions. Arundel: “You have the sense of distant warring factions in the city with the sound telling the player, ‘these folk firing at each other don’t concern you right now’. But then we very quickly need to swap to ‘look out, they’re firing at you’. The courtyard recordings all went into that layer of ‘thugs in the distance fighting’, while the recordings from the street and the end of the street went into the thugs firing at you. “The day in the steel mill was amazing – it’s quite a dangerous place and it took some organising to get permission but yielded so many incredible

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sounds. At one end the ‘cooked’ girders came out and a crane picked them up, dropping them off somewhere far off. Every now and again, from about half a mile away, you felt a massive thud and this distant metal clang would echo through all the warehouses. In part of the game, The Joker has set himself up in a steel mill but he’s half turned it into a messed-up funfair, hence also recording at the funfair museum in Devon. We were able to collect myriad ‘musical’ sound effects from old carousels and waltzers, as well as lots of mechanical sounds running on their own without the musical elements.” When it came to the sound of Batman himself, Arundel and his team strived to make the sound as believable as possible. “He’s got gadgets for everything and for me, they’ve got to sound totally believable – even when you know full well, in real life, the device couldn’t actually exist – it couldn’t do that thing – but it’s got to sound like it really could. John Roesch who did all the foley for Dark Knight and Batman Begins took care of a lot of those elements. Not that we wanted a copy of those movies – it’s a game and the way you approach it is different. We don’t have a Batman ‘audio bible’ as such but I think you just know when you’re doing it wrong. A good example is the cape opening – what we call the ‘wa-chonk’ – because that’s how it literally sounds as the cape flaps open and goes rigid. We spent ages on it this time out as you’re now outside for

60-70% of the time – high up, in tall areas. We really wanted the ‘wa-chonk’ to be a big pay-off sound-wise. As soon as you say that to a sound designer, you’ll get all these whooshy growls from animals – interesting, subjective content. But I really wanted it to be believable as a real world sound. It’s one of the sounds that people really care about.” The audio team’s involvement in dialogue recording can be summarised as technical management to determine consistency of microphone usage and signal paths, and ensuring everything is recorded faithfully and correctly from three different perspectives – using close mic, Lavalier (usually mounted somewhere on the actor), and shotgun. Arundel: “Probably 90% of the dialogue in the game is from a shotgun – more of a production sound. Recording this way means the actors can move around, waggle their hands, and do the things actors do. They are less bothered about mic position. However, with three feeds, you’ve tripled your decision making and file handling...” STEADY IN THE MIX

The final mix of music, sound, and dialogue was undertaken internally at Rocksteady’s two fully calibrated studios on a combination of Genelec and PMC monitors. Arundel did a first-pass pre-mix in 5.1. Then colleague Lee Banyard conducted a second pass, listening and tweaking Distributed in Australia by: Magna Systems and Engineering, Unit 2, 28 Smith Street, Chatswood, NSW 2067 Australia Tel: (02) 9417 1111 Fax: (02) 9417 2394

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He’s got gadgets for everything and for me, they’ve got to sound totally believable – even when you know full well, in real life, the device couldn’t actually exist

more in stereo. Lastly, a group review took place and the resultant mix was pushed out into the wider development studio where most of the art, coding, and design team listened on headphones or flat-screen TVs. The mechanics behind the mix process were down to Audiokinetic’s Wwise, the underlying audio middleware ‘engine’: “There are a tremendous amount of mix states – obviously one per room and then per type of game-play, be it combat, predator, conversation – hundreds of them. There was some debate about my initial mix. We’re always keeping an eye on loudness and I’m quite keen for the game not to be that loud that often. It frees us up to create some big surprises. I reviewed my initial mix at home and took feedback from other people, concluding we weren’t quite loud enough. I reduced the dynamic range at that stage and because we were using the McDSP limiter on the final output we didn’t have to remix the whole thing – just change the mastering limiter settings. Each mix iteration got easier but I think next time, we might bring someone in to do the final tweaking with completely fresh ears. “The hardest kind of aural perception test AT 44

I’ve ever done is having to write down four simultaneous music parts by ear. If you’re really good at listening, you can clearly hear four elements, but you’ve still got a hierarchy – you probably don’t listen that much to the bass, you pick up key notes and work it out. I think most people can hear two or three things at once before they start aggregating sound – lumping sounds together as say, ‘all the environment’ or ‘all the walla [background crowd murmur]’. When we’re mixing, we’re asking, ‘what’s the thing that leads this?’ A conversation? Music? Is it the sound design?’ This informs our mix. You have believable environmental sounds that signal to you the type of room you’re in with realistic reverb, but we also have other sets of sounds that are meant to push the player towards a certain type of feeling and these are not at all literal. We spend nearly as much time thinking about how the room should feel as to its sonic believability. “We use a fair amount of real-time DSP including a very realistic reverb and delay in the city areas which works well on the dialogue. You get some really nice moments as you run away from certain characters, the speech reflecting off the walls down the street. One game feature is a

surveillance scanner. When deployed, it surveils the thugs and tunes into their conversations. There’s McDSP FutzBox processing on that dialogue. As you approach the thugs, that cross fades into its furthest reflected and delayed reverb sound and then crossfades into a drier sound as you come up close – all mixed in real-time. Add to this what, for me, are totally believable city population sounds, and you really feel like you’re Batman roaming around Arkham City.” SUPERHUMAN EFFORT

Amazingly, and very unusually in video games, the same man who stands at the helm of this gargantuan effort, steering a global audio team towards their creative goals, also composed much of the music score, which weighs in at over two hours’ worth, recorded by Nick Woolage at Air Studios, no less. It seems that perhaps, something of the Caped Crusader’s superhero multi-tasking skills may have rubbed off on Arundel himself.

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GET OUT WHAT YOU PUT IN DIs: unglamorous? Perhaps, but it pays to know these audio workhorses a whole lot better. Text: Trevor Cronin

The humble DI box doesn’t receive a whole lot of attention. Certainly no one coos over them like a sexy valve condenser mic or an esoteric compressor. The truth is, the DI is the gatekeeper on a large chunk of your inputs and deserves your attention. In fact, get to know your DIs and you’ll be repaid handsomely with a cleaner more fullbodied sound. The core function of the DI is to convert a high impedance (and sometimes high level) signal into a low impedance balanced microphone level signal which is suitable for input into a microphone preamplifier. That’s the bare basics. But in the same way a microphone ‘converts acoustic energy into an electrical signal’ it only tells half the story. A-Designs, ART, Aphex, ARX, Avalon Designs, Behringer, BSS, Countryman, dbx, EMO, Hughes & Kettner, Klark Technik, M-Audio, Manley, MTR, Pro Co., Radial, Samson, Sans Amp, Summit Audio, Switchcraft, Line6, Whirlwind… there are more than 100 different DIs available, each with different features and different sound quality. So let’s explore some of the major differences. PASSIVE DI

These units sit at the budget end of the spectrum and require no external power to operate. Inside you’ll find some simple input circuitry with an isolation/balancing transformer on the output. More advanced models will have an input attenuation pad and an earth-lift switch. A good quality passive DI is a great interface to have in your system, perfect for use on keyboards or active guitars that have a high output. Like all audio gear, the sound you get out of them depends on the transformer and good transformers (like the industry favourite Jensen brand) are expensive. Some dirt cheap passive DIs are genuinely awful and prone to pick up noise from being near other equipment. Be careful, a bargain may bite you when you are not watching. Pros:

✓ Simple passive device ✓ Can sound good ✓ Robust. Cons:

X Not suitable for low level inputs.


These are the most common type these days. Suitable for all types of inputs, from the lowest level (such as passive pickups in acoustic guitars), through to full line level from effects units. Some can even accept speaker level input, which is a very handy feature. The better units also have a female XLR connector for input, giving it the dual purpose as a problem solving isolation box. Inside you’ll find active circuitry, requiring power to operate, with an amplifier circuit as well as input attenuation and perhaps some filtering. The output is generally electronically balanced, however, the better units still retain a high quality transformer on the output. You get what you pay for with these devices, a cheap active DI generally will sound that way. Pros:

✓ Able to accept a variety of input levels ✓ Input filtering Cons

X Comparatively fragile X Requires power X Can sound ‘harsh’ ACTIVE MULTI-CHANNEL

These rackmount professional models are designed for large keyboard or multi instrument setups. The quality is often very good and some can double as a high-quality monitor mixer. They save a lot of on-stage spaghetti and simplify and speed up the setup time. DI WITH BENEFITS

Some units have high- and low-pass filters or a simple tone control included. Many amp simulator pedals can also double as a DI box. The Line6 Pod and the Sans Amp models have this feature and work well. Saying that, check the sound quality before the show or recording session and have a regular DI handy, just in-case. DIs customised for ‘audio visual’ use have RCA and or minijack inputs – so no adapters required to plug in the client’s iPod. The Hughes & Kettner Red Box connects to your electric guitar rig and gives a good amp sound without using a microphone. AT 47

The Avalon U5 – more customarily a reclusive studio dweller – shows its touring credentials.

There are some DIs that can also be used backwards (line in/instrument level out). This is for sending a recorded DI’s track into a real amp so you can mic it up and record that sound. It’s also great as an effect (over-driven Marshall on your lead vocal anyone?). Re-amping lets you email your guitar track to that friend interstate who has that wicked vintage amp collection. The Littlelabs Redeye DI is a popular choice for this use. MONEY NO OBJECT

TREV’S GRUMPY GUIDE TO ACTIVE DI SETUP 1. Preset the DI box with the gain or pad switch set to minimum, earth-lift off and EQ/filters off. Are the internal batteries okay? Are the connector contacts clean? 2. Turn down and mute your microphone preamp/ input channel and flatten the EQ. Patch your DI into the system, positioned somewhere safe from punters, liquids and foot traffic. Apply phantom power if needed. 3. Set the instrument level to about 75% (allowing the player some 25% extra, as they will like that option). 4. Preset the mic pre gain with pad out and channel muted to about 50% (12 o’clock on the gain pot). 5. Ask the muso to play at ‘performance level’ one of the tunes that they’re about to perform. Have a listen to what it’s like on stage as well. 6. Set the gain or pad switch on the DI to get somewhere near the correct level at the mic pre. Get the lighting guy to stop flashing the lights in your eyes while this is happening. 7. Make some fine adjustments to the mic pre gain to set the correct level (use the pad if needed), somewhere below 0dB, as the player has that extra 25% they can use. 8. Feed the signal into your speakers/monitoring system and assess what you’re hearing. Tell the other players to shut up so you can listen. 9. Use the DI’s earth lift to remove any hum. Use the instrument’s EQ, DI filter/EQ to get a sound that you feel, together with your muso friend, is okay to work with. 10. Work some of your sound engineer magic: some compression generally works well with active DI inputs and maybe some reverb. Add to taste.

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If budget is not an issue, but sound quality is, here are three of the best to look after your money channel: Avalon Designs U5, Summit Audio TD-100, Manley Tube Direct. These units are very expensive, with specifications and sound to match the price tag. They are used by the top professionals. BUILT-IN DIs: A WORD OF CAUTION

Many stage amplifiers come with a ‘DI out’ and this may work fine most of the time. However, do they supply the correct mic level? Do they have an earth lift? An isolation transformer? Do they give a signal split before the amp electronics, so you can mess with with your stage sound without affecting the feed to the PA system? Not sure? You’re better off using a good ol’, regular DI. TALES FROM THE REAL WORLD

DIs are such handy devices, helping me out of a whole variety of problems. The DI allows you to take a feed from various parts of a signal path – this can get you out of some scrapes. Patched in before a guitar amp a DI can help avert disaster if the stage amp blows up – just feed the DI signal into the monitors and keep the show going. Set this up beforehand with some EQ and even FX to get the right amp sound, so you have a Plan B ready to roll. This happened with one of my favourite guitar players, Dion Hirini, at a major album launch I was mixing – a genuine show stopper was averted thanks to the DI and some preparation. You can patch a DI both pre and post stage effects (and even post amp with the right DI unit) allowing you to blend all with the microphone signal (or not) to get the killer sound

in your mix and have an emergency backup at the same time. I recall years back receiving a panic call from a St Kilda studio which was recording the orchestra parts for the film Elizabeth. The problem? Two consoles patched together = one big earth hum. I arrived at a very crowded and sweaty studio with my trusty ARX DI-6, patched it in between the consoles, pushed the earth lift switches and the problem was instantly solved, allowing a very expensive session to proceed. I forgot to invoice for that… it’s probably too late now. Then there was the time I was mixing monitors at the Glastonbury Festival in the UK for one of the headline acts. We’re allocated 15 minutes to set up the sound before the show, and after 14 minutes we find a big problem with all the DI channels from the sequencer (which was patched into our own rackmount DI). In desperation I decided to run across the stage with an armful of BSS DI units and do a quick re-patch, run back to the monitor console, push six phantom power buttons on the console, and bingo, the show started successfully in front of 80,000 people. Phew. Keep your DI boxes happy. Keep some fresh batteries, so you don’t have to always rely on phantom power; do a comparison sometime between running on batteries and using phantom. Contact cleaner (my choice is the aircraft-grade CAIG deoxIT) clean all your instrument jacks and sockets as well as DI inputs and outs, this can make a big difference to the noisefloor. Look after your DIs. Keep a padded waterproof/dustproof case to protect your investment. After all, if you buy good quality DIs they should outlive you. Trevor Cronin is an enormously experienced sound engineer who loves his ARX DIs. He also packs a BSS AR116 battery-only DI (which was the industry standard until replaced by the AR133), and an Avalon Designs U5 – Trev’s newest and most expensive DI and “it sounds wonderful”… “your clients instantly know you take their sound seriously when you plug them into this”.

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KEEP IT CLEAN Having some restoration plugs can save your skin.

Okay, the drums are sounding great, the guitars are all plugged and exciting, and the vocalist is back from the café. Time to record!

Text: James Roche

Fast forward to the next day, as I listened to the mega-hits-in-the-making… What’s all that noise? Where is that hiss coming from? That annoying buzz? There’s a hum in there too! And the vocalist sounds like she’s running a saliva factory – clicking and crackling through the whole take. I can’t use this! It’s moments like these when some noisereduction software can come to the rescue. Plug-ins and standalone apps from Waves, Sony and many others are designed to get rid of those pesky non-musical audio artefacts – and I can testify to how marvellous the results can be. Recently I used Izotope’s RX2 noise-reduction plug-ins to help me turn a whole album’s worth of basketcase multi-tracks into noise-free bliss. BROADBAND NOISE: DEATH BY VALVE

RX2 packs a number of task-specific plugs. The first one I needed was Denoiser, which removes broadband and tonal noise that remains largely constant throughout a recording. I had a percussion mic that, unnoticed on the day, was producing a lot of white-noise. It was the only mic on this particular instrument, a djembe, and there just wasn’t enough spill of djembe in the other mics for me to mute the noisy track. It sounded like a valve with one foot in the grave! The band didn’t play to a click, so I couldn’t easily overdub one instrument of a percussionist’s entire performance. In other words, I had no choice but to use this track.

RX2’s Denoiser for broadband noise; Declicker and Decrackler (right, opposite) for the likes of vocal spittle, Declipper to salvage the odd ‘over’.

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Noise reduction plug-ins work by ‘learning’ a noise profile. If you can isolate just the noise and play that into the plug-in, the software can go to work on the rest of the track removing that noise profile from the music. At the front of this particular song before the count-in there was absolute quiet from the band so I could isolate the noise. From there I could let Denoiser weave its magic. It worked a treat, turning a crisis into salvation.


I might have reduced the audible noise from the dodgy valve but I could see on the spectral display of the plug-in that there was a lot of energy down in the subs, below 40Hz. I couldn’t hear it, but I could certainly see it – it was the loudest frequency band on the track! The likely culprits would be air conditioning and distant traffic. Usually when I record, I roll off the subs on mics for anything that isn’t specifically a bass instrument or a large drum. However, I hadn’t engineered this particular session, and there was deep dark energy on every mic. Luckily, the Denoiser plug-in handled this job very well, and at the end of the processing I had clean tracks with no rumble and no wideband hiss. GUITARS: RATTLE & HUM

Next problem – guitars! On this album we used a really old-school guitar, with noisy pickups. The recording had constant hum from both the amp and the guitar, and occasional high-frequency buzzing when the guitarist wasn’t actually touching the strings to earth the instrument. I applied a combination approach, using the Denoiser for the buzzing and the (aptly named) Hum Removal plug-in for the constant AC noise. Hum Removal has plenty of parameters to play with but I found I was able to achieve my aims by changing the setting from 60Hz to 50Hz, to match the frequency of power in Australia. Like magic, the hum disappeared! I was beginning to love this software. It was saving my bacon. VOCALS: SPRAYING IT

Time to deal with the vocals. They sounded pretty clean with no real noise or hum problems – but that saliva! It’s in the singing; it’s in the breathing – it’s everywhere! Click, crackle, splurt! I usually zoom right in and redraw the waveform to get rid of a random click – but this was just too large a task for all this crackling. I needed another solution. I turned to the RX2’s Declicker and Decrackler. Their job is to remove very brief audio artefacts that aren’t like anything

Hum Removal: great for excising AC noise (just flick the Frequency Type from 60Hz to 50Hz!) or pickup hum.

else around them. This is more difficult in a vocal where there are desirable sibilants and transients right alongside undesirable clicks. Saying that, I found that with a bit of careful massage I could remove 90% of the saliva problems by using the software, which was a major time-saver. On another track I also found a use for the Declipper plug-in, which turned a number of moments of slightly distorted vocal back into something useable. PAID FOR WITHIN A WEEK

This was the first time I’d ever needed so much noise reduction and restoration processing on an album, but I reckon I’ll never leave home without it! It saved me time, and saved the project money by avoiding rerecording. It paid for itself in the first week, and it isn’t often you can say that about a purchase. Mixing the album was now a pleasure, as I could turn everything way up, get it all forward and present, and the end result was very intimate and up-close. There is no way I could have achieved that result with the noise in there. ‘Keep it clean’ is my new motto!


Head to the AT site to hear James’s before and after experiences with the RX2 suite.

Australia-wide delivery Ph: 02 9283 2077 Fx: 02 9283 1337 265 Sussex St, Sydney 2000 AT 51


MEDICI STUDIOS Good things happen when the rich channel their altruistic streak into recording music. Text: Greg Walker Photos: Corey Sleap

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The Medicis – a filthy-rich Florentine renaissance dynasty – were famous for their patronage of the arts. If you were a penniless muso or artist then there was no better way to pay the bills than get a Medici commission. 500 years on, it’s still no easier for modern minstrels to get a leg-up, and generous patronage is just as important.

oversee the installation.

Medici Studios is a purpose built facility in the Geelong area of Victoria, established by millionaire owner Paul Morgan to assist young musicians by providing an affordable worldclass recording space. It’s the same traditional rationale at work: young talent needs money and experienced assistance to realise its potential. Medici Studios is providing just that.

GW: Carter Rollins is a case in point. It’s unusual for a local band to be so ‘big in Japan’ with credible yet comparatively modest success in Australia.

Plenty of talent of all persuasions comes through Medici Studios, but its become particularly well known for its Surf Rock, and Medici has found a very receptive market in Japan. Carter Rollins are the current poster boys for Medici’s push into Japan, and are a shining example of what’s possible with decent gear, a real studio, technical know-how and some music business connections. AT spent some time with house producer/ engineer, Michael Stangel, to learn more about the Medici Method. RE-RENAISSANCE

Greg Walker: How did the relationship with Medici Studios start? Michael Stangel: I met Medici’s owner, Paul Morgan, around four years ago. I’d heard about this bloke in my home town of Geelong who had built this amazing studio and through a mutual contact I arranged a meeting. I could not believe the setup and how much effort had gone into the design and the equipment – I was in shock. After talking to Paul I got a better idea about what his goals were and it seemed like a perfect fit. He originally designed the studio to be a place where young musicians could have a place to record. Paul wanted it to be a state-of-the-art facility and brought in [Australian producer] Chris Scallan to

Once we’d agreed to team up I started working on some projects Paul had been stockpiling and started getting them completed and out into the marketplace. With my background in A&R we began to find ways to produce the music and gain revenue for the artists through overseas markets as well as locally.

MS: I have previously recorded an album with them, but this will be their first release with the help of a major record company and I’m really excited about their future. They are signed in Japan and will be moving to the USA early next year to have a crack at the Yanks. Carter Rollins is comprised of Justin Carter and Johnny Rollins [pictured in this story completing an EP with Michael] and have the most unique blend. Justin is a real fan of acts like Neil Young, and Johnny is more into bands like Fall Out Boy. The blend is amazing and I’m excited about the reaction to the new EP and album.

The guy screamed out, ‘That’s it, that’s it, that’s the sound!’ To my amazement they wanted to hear digital clipping

GW: Talk us through the usual setup you use when tracking a band like Carter Rollins. MS: With most bands I generally start with pre-production. This is a time for the band to bring in their songs; we’ll play them through live and then talk about the vision they have for the production. They would generally have more songs than we need for an album so we can have a choice of potential tracks. With Carter Rollins we generally add Wurlitzer, Hammond and bass to the recordings, as they are a duo for the most part – they’ll get session musicians to come in for the larger shows. I get to play all the extra instruments, so it’s lots of fun for me. After we figure out the tracks to record we get down to business. Johnny plays drums and has a very aggressive style, so I like to mic the room well. I use a

WORKING WITH THE SSL AWS900 Michael Stangel: The SSL is amazing! I like the sound of the AWS900 and the G-EQ option really adds variety to my mixes. However, I do a lot of mixing in the box now. I was previously never a big fan of doing that but on my last trip to the USA a good friend of mine, Bill Appleberry (Stone Temple Pilots, Puddle Of Mudd, Wallflowers) let me in on a few secrets to the art of mixing in the box and I have not turned back. I have always liked SSL consoles and have worked on the G, E, and K series boards. I must admit my favourite desk to track on is the beautiful old Neve 1081 at 301 Byron Bay. It’s the old Festival Studios desk and there is definite romance going on between us. My favourite place to mix is on the SSL K at 301 in Sydney.

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Neumann M149 that I put in the kitchen around 15 feet away, which gives me that roomy Led Zeppelin sound and also two Neumann U87s about five feet from the kit. I also use the good ol’ SM57 directly above the kit. The next step will be guitars. Justin is very old school, so we like to get as many valve amps as we can find and start having fun. I’ll get three or four amps and put an SM57 in front of each with one Royer ribbon mic placed equidistant from all of the amps positioned in a semi-circle. We then use a multi-switch to try lots of combinations of guitars and amps until we get the right sound for the track. From there I will record the extra instruments and vocals. SURF ROCK’S POOL OF YOUNG TALENT

GW: Any other gems you’ve uncovered?

RECORDING BANDS & THAT MAGIC TAKE MS: It doesn’t matter what trend is in at the time; great bands with a unique sound will always find success. And they don’t always have to be the best musicians the world has ever seen, but they have a blend of technique and sound that makes them unique. If I find that, my job becomes easy – just record the session faithfully and don’t get too tricky. A very good friend of mine and one of my mentors David Nicholas would tell me stories of working with bands like Midnight Oil, INXS and The Pretenders. He always told me to be ready at any given moment for the ‘magic take’, and that the trick was to keep focus and know when the magic take just happened. With bands where I know that I am working with musicians who can focus, I will always try and get live takes. Depending on the studio, I will isolate the drums as best as I can and then it’s a matter of waiting for the magic to happen. I always try to keep the recording path as unaffected as I can. I might boost some 60-80Hz in the kick and maybe slam the bottom snare mic with an LA-2A limiter, but other than that it’s down to tuning the kit and placing the guitar mics according to the sound I want. As far as bass goes, I always run a DI and then usually an [Electrovoice] RE20 mic on the bass amp. AT 54

MS: I’m always working on new sounds and looking for the next thing that will excite me. For example, we have two young girls who call themselves Lash 78. They are only 12 and 14 years old. They have blown me away with their confidence and natural talent. GW: How do you approach working with young talent like Lash 78 as opposed to more seasoned artists? MS: Generally with young talent I try not to say the word ‘no’ or discourage them from trying new things. I remember an occasion when I had a young band in the studio with an average age of around 16. They wanted a certain kind of distortion on a particular lead break. So I began to break out the usual suspects: my Sans Amp and a few other toys including Amp Farm but I couldn’t get the sound they were after. I remember plugging in the amp mic and forgot

to turn the trim down on the channel which was clipping the ProTools input. The guy screamed out, ‘That’s it, that’s it, that’s the sound!’ To my amazement they wanted to hear digital clipping, and after frowning and shaking my head I listened and realised ‘you know what, it sounded perfect for the part’. I was quickly reminded that you have to be open to unconventional approaches, no matter how wrong they might seem. As for the Lash 78 girls, they’re like seasoned pros and they have no problems with any of my suggestions. In fact, they often get frustrated with my tact. Compare that to some seasoned pros, who shall remain nameless, who are precious beyond belief! GW: Finally, is the focus on surf music strategic or just serendipitous? MS: It’s something I fell into really. The studio is located close to Torquay and there is a lot of young talent here that just happens to love the surf culture. A lot of those people come through the studio sooner or later – much like Carter Rollins. Kym Campbell is a Sydney native originally from the USA who is also making great progress in the surf music community. After spending time with artists like Justin and Kym, I began to get a real feel for the more earthy tones they needed to get their songs sounding right. I have really embraced a lot of vocal harmony and percussion sounds that help give the music a real coastal flavour.  You can check out some examples of Michael’s work and see pictures of the studio at www.

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WHAT’S ON Trackdown Digital and Paul Pirola announce the creation of a joint business offering a complete audio and music post solution for film, television and interactive projects. The business will be called Sound & Music Solutions. Paul Pirola, one of Australia’s leading sound designers with such credits as The Taking of Pelham 123, Sanctum and Knight & Day will bring his wealth of experience to the new venture, while Trackdown Digital with music production credits for Happy Feet (I & II), Moulin Rouge and Knowing will add their skill set in clever music solutions, production management and, of course, their world-class scoring stage. Sound & Music Solutions will initially operate in both Sydney and Melbourne. The combined resources and facilities of both Trackdown and Paul will be at the disposal of the new venture. go: www. for more. The Grove has unleashed the power of Studio 3. It’s a hybrid mix room for Grove engineers only. With a mix of analogue and digital gear, Studio 3 has already proven to be popular, with 2012 dates already filling up. The Grove has played host to Maddi Saward, Drawing North, St Leonards, Shinobi and Shortstack. Two albums recorded at The Grove this year were nominated for an ARIA: Eskimo Joe and Josh Pyke – nice work chaps. Damien Gerard Studio is all go. Rose of York

have been cutting some great vocals with Producer Paul McKercher – there’s plenty of love for the vocal chain set up through the API lunchbox – Chris’s voice sounding tops. The Blake Saban Three made the trek from Tamworth to record their debut album over eight days with Russell. They played a slot at the Sydney Blues Fest while in town. Bob Pigott finished all his mixing and ended up with a superb sounding album – tracked at Abbey Road London; overdubs and mixing at DGS with Russell Pilling; mastered by the amazing William Bowden. DGS regulars, Physic Asylum from Canberra have been back in the studio with a new lineup to do more tunes. Calling Mayday have completed a very productive preproduction night and are now working hard at rearranging and honing the songs to be ready to track the new tunes early December.Other sessions have included more work from Decline of the Reptiles, Michael Peter Daxton, Alex Chudnovsky (Classical), Musk, and plenty more. ‘King’ William Bowden has been soaking up the accolades for his mastering of Gotye’s worldbeating album, Making Mirrors. Lanie Lane’s latest has also had some King Willy Sounds’ sonic fairy dust, as has Maple Trail and Simon Starling (produced by Steve Kilbey from The Church). Red Door Sounds is the Melbourne recording

studio co-owned by Rae Harvey of Crucial Music, producer Paul ‘Woody’ Annison, Chris Cheney (The Living End) and Emma Cheney. Red Door recently acquired Sydney studio Big Jesus Burger’s coveted Neve 60-channel Vatican desk. This amazing piece of equipment will be up and running in the coming months. Powderfinger’s Darren Middleton will also be working out of Red Door. Darren is keen to work with fellow artists in a producing and songwriting capacity. Children Collide commenced recording their third album with Woody at Red Door. And lately the studio has also played host to The Living End, Lanie Lane, City Riots, Hunting Grounds, Celadore, Rockets, Lee’Mon and more. Sing Sing’s outboard collection is one of the most formidable in the country, and its stocks continue to soar with the addition of an Empirical Labs Fatso and an Avalon 747 – both added to the M Room racks. The coup de grace is a new Tubetech CL1B, and this will be available as a hire item or you can throw yourself on the floor and have a toddler’s tantrum and (maybe) get it thrown in as bonus gear on your studio project. Coming soon to the K Room is a Shadow Hills Mastering Compressor. Four of Sing Sing’s Neumann mics (3 x U47 Valves and 1 x U48) and their power supplies have just undergone a complete ‘makeover’ and are fresh back from their trip to

Fishtank Recording Studio has become the new home of s:amplify, an all-in-one music production house. Between the collections of the two in-house producers, Josh Abrahams and Davide Carbone, the equipment range is so extensive as to cover almost anything you can think of. Rocket-fast eight-core Mac Pros with pretty much every soft synth and plug-in: ProTools HD3, NI Komplete 8, Logic Pro 9, Ableton Live 8 Suite, Omnisphere, and the IK Multimedia Total Studio bundle, just to mention a few of the biggies. s:amplfy has most classic electroacoustic keyboards you can think of (and some you can’t) as well as three valve Hammond organs, two Leslie speakers and a Steinway grand piano, not to mention a pile of great guitar amps and guitars, a huge range of classic mics, ranging from early ribbons through to classic valves and sharp new small diaphragms, and a collection of vintage preamps and compressors that needs to be seen to be believed. FISHTANK RECORDING STUDIO AT 56

The studios at s:amplify are used

primarily as composition tools by Josh and Davide, where they work on a wide range of projects covering most aspects of the music industry, including: co-writing, remixing and producing for artists such as Carl Cox, Kaz James, Guy Sebastian and Moby; creating sonic identity and music packages for clients such as FutureBrand, Imagination and Boffswana; consultation, design and synth presets and patches for instruments including KDJ1, NI Massive, and Rob Papen Punch; and composing, producing and mixing TV commercials for clients including Tourism Australia, Dove and the Oprah Winfrey Network. And, yes, s:amplify also records bands; some of the local faves to come through have been the Sure Shot Hunters, Robyn Loau, and The Genie. The s:amplify motto is ‘Craftsmen of Authentic Sound’, so you can be confident that these guys will craft something unique, specific to your needs.

Adelaide where Rob Squire treated them to the equivalent of a microphone day spa treatment. All mics (now shiny and bright and thoroughly fit for service) are already in use on sessions. The calvacade of primo talent to darken the Studios 301 doors includes: Cameras In Your Room - Produced by Michael Morgan; Deep Sea Arcade Girls – Mastered by Steve Smart; Marvin Priest Feel The Love – Mastered by Leon Zervos; Cold Chisel All For You (Single); Zowie’s optimistically titled Smash Hit – Mastered by Andrew Edgson; Mark Vincent Songs From The Heart – mastered by Leon Zervos; Shannon Noll A Million Suns – Mastered by Leon Zervos; Matt Corby Brother – Produced by Tim Carr and Mastered by Andrew Edgson; Catcall Satellites – Mastered by Andrew Edgson; Washington Insomnia, and many more. Also: Ben Feggans joins 301 Mastering. Ben’s somewhat of a local legend in the Indie/ Dance, Electronic, Folk, Dub and Hip-Hop scenes, with a mastering discography including Flight Facilities, New Navy, Deepchild, The Versionaries and Flatwound.

hosted some great freelance engineer talent with Greg O’Shea, Lachlan Carrick and Nick Huggins and also Casey Rice recording Brian Chase of The Yeah, Yeah, Yeahs. Blair Joscelyne from Nylon Studios has been busy writing music for McDonalds USA, Toyota, VW and Visa this month. He’s also been playing with some new home-made synthesisers and circuit bending any toys found left laying around the studio. His track for Cascade Beer The Feel was nominated for Best Music in a Commercial by APRA this year, and his new single Collide Featuring Erin Renee is out now on iTunes. It’s been hectic at Mixosmosis, Nathan has been in lockout mode mixing the new Citizen Dog Album, due for release in the New Year, and putting the final touches on Brett Hunt’s new album, also due for release in the New Year. On the post production side off things, Nathan has been busy sound designing and mixing the short film The Republic Of Simon by local Sydney writer and director Eddy Bell, and next on the slate is sound design and mix of director Jake Weisz’s short film Pride. Gigs in between include some recording and mixing with Sydney rapper Benzy.

Tender Trap Studios in Northcote, Melbourne, has been quietly going about the business of making noise. Coming in to hang out at DOMC of late has a been a stack Recent albums include the new and beautiful Tex Perkins of incredible music. James at Heliport studios came in and the Band Of Gold release (recorded and mixed by with several projects and his SSL Duality makes them all Roger Bergodaz) and a rock and soul album featuring the sing – that and he has great ears. Robbie Brock came in to exceptional talents of The Rockwiz Orchestra (recorded by do some vinyl transfers of classic Australian bush ballads; AJ Bradford). Other projects include Tracy McNeil, Sime Bec Whitehead’s debut EP has revealed “an amazing voice”; Nugent and The Wolfgramm Sisters and a great session with His Merry Men came through for some DOMC love; Jason Rob Snarski (The Blackeyed Susans) with Dan Luscombe Delphin was back with more dance tracks – “his first ep (The Drones) and Dave Williams (Augie March). Tender is going to blow minds”; Dan Sugars is back in for some Trap has also been home to some soundtrack recordings DOMC time as well. An album of electronica came in of late: Dan Luscombe completed the tracking and mixing for SBS documentaries from Allan Gilmore; and to top of his compositions for the first series of Laid (ABC1) and it all off Dom got himself my new JLM Audio VUPPM studio owner AJ Bradford continues with various music LED meter – “Joe had it custom made to slot into my to picture projects, last year winning the AFI Award for Crookwood console”. Best Sound In A Documentary for his score for the ABC Ear Monitors Australia #44 1/12/05 1:47 PM Page 1 Jim Moynihan from Spoon Studios has recently produced documentary Inside The Firestorm. The studio has also

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a new theme for Foxtel’s Toon Time Looney Tunes show, as well as the sound design for a Japanese Puma ad. Also recently completed was the mastering of the new album by Melbourne roots group Simmer, plus the debut EP by Composite. New studio sonic improvements include 12 custom made bass traps, and a large cloud. Other studio purchases include a Haken Continuum Fingerboard and the Dangerous ST monitor controller. Bob Scott has been recording a CD of old Australian Bush tunes with Warren Fahey for ABC Music. He’s dabbled in some FOH sound for Voices in the Forest Concert at the National Arboretum in Canberra, with mezzo Anne Sophie von Otter and a Canberran Orchestra. In New Zealand Bob’s been recording traditional instruments of the Maori for a dance work by Victoria Hunt, using a borrowed Sound Devices 702 – “a beautiful little location recorder” – with a pair of Sennheiser MKH20s ‘dodgily’ mounted in a borrowed Rode blimp – “a very effective little location rig”. To round out a busy month or so for Bob, there were single mixes for Jared Underwood’s Batterie. Sound design for Ensemble Offspring and Theatre Kantanka’s Bargain Garden at the carriageworks theatre. Music editing for Nigel Westlake. Mixes for On the Stoop; recording a wind quintet of Lyle Chan’s Rendezvous with Destiny with narration by Bob ‘Not Quite as Dodgy’ Carr. Sound design and some composition for a kid’s show “about Dinosaurs… lots of live pitch shifted growling fun”. Finally, congrats to His Dodginess for picking up an Australian Screen Sound Guild gong (with Doron Kipen and Sophie Raymond) for recording the music on Mrs Carey’s Concert. Deluxe Mastering has been powering into the festive season like a freight train. Tony ‘Jack the Bear’ Mantz has had a busy time putting the final lustre on projects for the Dirty Three’s latest album with bonus tracks, Canadian rockers Bel Riose, Hex, hip hop outfit EMR, a live DVD release for Engine Three Seven, Shadowgame, Deja, Ash Grundwald’s new sideline project, 46 Clicks, Garcon Garcon, James Frew, Bankrupt Billionaires, Ben Taylor, and dance releases for US-based Royal One Records. Projects mastered by Adam Dempsey include new and upcoming releases for Sons of Messengers, Jimmy Dowling, Playwrite, The Pirates, Taylor Project, Blue Shaddy, Adam Eaton, Selwyn Cozens, Tom Strode, The String Contingent, Mr Jimmy, Jason Delphin, Andy Jans-Brown, and this year’s NMIT songwriters’ compilation, part co-produced by Greg Arnold. s:amplify has had a very busy year working on the epic orchestral composition, arrangement and production, sound design and implementation for the live immersive Alienware launch event which went out via youtube to a worldwide audience. Carl Cox’s new album All roads lead to the dancefloor, co-written and co-produced by s:amplify, was released in June to rave reviews. Josh and Davide have been appointed music directors for Sydney NYE, creating the soundtrack for both the 9pm and midnight fireworks shows. Their presets pack for the NI synth Massive was one of the top sellers on the Loopmasters website, and they are putting the finishing touches on the feature set, sound design, presets and demo loops for a hardware synth, KDJ1, being built by Japanese company Cyberstep. Recent gear purchases have been Ohmicide and NI Komplete 8 on the software front, and a Lowery Wandering Genie organ from the seventies. Megaphon Studios had Dune Buggy (ex Wolfmother drummer and bass/ keys player) for some tracking sessions, Paul Mac spent time with engineer Jon Boy Rock to lay down some beats and general tracking, Ashleigh recorded some folk tracks with engineer Chris Hancock, On the Stoop brought its Gypsy jazz vibe to Megaphon with the help of engineer Shane Fahey, rock experimentalists Galaxy Fuzz Band also did some tracking with Fahey, as did Steven Ray Dan and James Domeyko. There’s plenty happening in December as well, with Ungus Ungus Ungus, Nic Cassey, Escape Syndrome, Dave Symes, Mat Wicks, Jozz Scott, Sri Diger and Mike Berkley all clocking on at Megaphon.



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What’s On wishes everyone in the Australian studio firmament a very happy Christmas… and a prosperous year of month-long lockouts in 2012!

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PC AUDIO Windows 8 looks to take on iOS and Android, along with speeding up your desk/laptop. But what about audio? We go Windows shopping. Text: Martin Walker

Described as “a re-imagining of Windows from the chip to the interface”, it’s about time we started talking about Microsoft’s Windows 8, especially since indications are that it may change quite a few aspects for the recording musician when it’s released sometime in 2012. While the largely familiar Windows interface remains, a new touch-based version (borrowing heavily from the Windows Phone) is also simultaneously available to users – Windows 8 will be a single operating system for both traditional desktop/laptop PCs, the new Windows-based tablets, and even mobile phones. The first tablet computer running Windows 8 has already appeared, from the huge mobile phone manufacturer Samsung Electronics – it unveiled a new computer model running a W8 beta version at the Microsoft BUILD developers’ conference in California in September. Prior to this, Samsung relied heavily on Google’s Android operating system, so let’s hope that this heralds the start of wider acceptance of Windows 8 for future tablets – we need some stiffer mainstream competition for the ubiquitous iPad! A ‘Windows Store’ similar to Apple’s App Store will be introduced to partner Windows 8 with downloadable apps from Microsoft and other developers, so hopefully we can expect plenty of new music PC-based Apps as well!

Mouse (responding to horizontal thumb swipes and one, two and three-finger gestures) shows how those still using traditional screens will be able to join in the fun. This emphasis on scaling, along with Windows 8 system requirements remaining identical to those of Windows 7 (a minimum of a modest 1GHz processor, 1GB of RAM and 16GB hard drive space) shows a new trend – ‘ease of use’ rather than ‘more raw power’ – and ties in with studies that suggest we will be moving towards virtual and cloud computing architectures in the next five years. Cloud computing, with on-demand network access to a shared pool of computing resources could make perfect sense for a lot of musicians who already store audio files and collaborate online with their peers. Perhaps we’ll see ‘band’ as well as individual or corporate licenses for software so that all musicians in a group can legitimately share the same on-line apps. Now that many of us already have more than enough personal computing power, a new easier to use shared environment could make a lot of sense! PUTTING THE BOOT IN


Sticking with ‘ease of use’, it looks like we can expect significantly faster bootup times from Windows 8 as well, even compared with the improvements found on Windows 7. Using a new hybrid shutdown and boot process that combines a normal cold boot with a modified ‘resume from hibernate’ mode, only the operating system kernel processes get hibernated, but not applications or user data. The resulting hibernation data is therefore much smaller, and Microsoft claims boot times will be 30 to 70 percent faster than with Windows 7. Those with Windows 8 installed on solid state drives can apparently expect their PCs to boot up in around eight seconds – just the ticket when inspiration strikes!

Microsoft also says “a Windows 8-based PC is really a new kind of device, one that scales from touch-only small screens through to large screens, with or without a keyboard and mouse”. So, apart from dedicated tablet computers, multitouch trackpads and touchscreens are going to be supported, while the recently released Touch

For those on the move, the new ‘Windows To Go’ lets you install Windows 8 on a USB flash drive and boot from it, complete with your personal user settings, applications and files (a USB 2.0 drive will work, but USB 3.0 speeds will be hugely preferable). In theory, you could install your favourite sequencer and associated audio plug-

Also described as “fast, fluid and dynamic”, Windows 8 certainly features Microsoft’s most uncluttered interface to date, abandoning grids of occasionally confusing icons in favour of a larger, cleaner and easier to read ‘Metro-Style’ start menu interface containing lots more text, a horizontally panning layout and a mosaic of large animated tiles – one to launch each of your main applications as well as indicate their current status.

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ins along with your latest ‘tracks in progress’ on the same flash drive (space permitting), wear it round your neck on a chain while travelling, and then plug it into other Windows 7/8 PCs at your destination and booting into your own customised Windows 8 environment. This could be safer and easier than carting a notebook/tablet around all the time, and feel rather like having a custom steering wheel for someone else’s car! However, in practice I suspect your sequencer may object to finding an entirely different audio interface plugged in when you use ‘Windows To Go’ on your mate’s PC, especially if it has a different number of inputs and outputs. A NOTE OF CAUTION

I’m certainly looking forward to trying out all these new features, and am far more excited about Windows 8 than I was about Windows 7 (which was really Vista MkII with the majority of bugs fixed). However, while being able to run Windows 8 with ‘less than 300MB of RAM’ or a modest CPU is wonderful for those carrying around mini-tablets and mobile phones, let’s hope these ‘resource-reducing features’ won’t backfire on those musicians who want to install and run it 24/7 on PCs running 8GB or 16GB of RAM and steaming 24-bit/96kHz audio using the latest cutting edge processor. In the past, many CPU throttling, RAM, battery and other power-saving schemes have resulted in insurmountable audio glitches for the studio, particularly at low latency settings, because we have very different computing requirements to most other users. Moreover, despite making the largest amount of noise, musicians are still a very small minority of PC users in the grand scheme of things, so our frustrations get rarely heard. On a lighter note, Windows 8 will finally see the end of the dreaded ‘blue screen of death’ crash message. Sadly, that doesn’t mean crashes are a thing of the past – it’s going to be black instead!

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MAC AUDIO Sleep or Hibernation? Brad discovers an assault on his battery. Text: Brad Watts

Recently I had a bit of a hiccup with my now ‘past-itsuse-by-date’ MacBook. It’s an early 2008 model packed to the gills with RAM and hard drive space. It chugs along at 2.4GHz, it has a Firewire port, and I like it. However, due to a recent change in vocation, I’d left the good ol’ MacBook sitting around doing nothing for about a month. It had gone to sleep and, as is the standard operating procedure with relatively recent MacBooks and MacBook Pros, had used whatever remaining battery power it had to keep the RAM contents intact, waiting in vain until I opened the lid and let it all spring back to life. Unfortunately that didn’t happen. The computer said ‘no’. No biggie I thought. I’ll plug the machine into the mains and recharge the battery. To my joy, the MacBook booted as expected. Yet to my disgust, the battery menu told me the battery wasn’t charging and required immediate replacement – bugger. I attempted all the tricks; zapping PRAM, resetting the PMU, permissions, a clean operating system, but nothing resuscitated the battery – it had been keeping the RAM contents alive for ages and completely depleted its charge doing so. This is the death-nell for MacBook batteries. I was left no choice but to buy another battery – damned if I was paying through the nose for a battery through Apple itself. This was a job for eBay. Within 10 minutes I’d ordered a replacement for the princely sum of $38. Not a bad deal I thought, and kinder on the bank balance than the $149 genuine article… until I fitted the battery. But that’s another story which I’ll get to shortly. In the meantime, how does one avoid a battery completely discharging and performing the Apple Seppuku? There are three modes of sleep with Intel-based Apple laptops (and desktop machines, but it isn’t so much an issue with a desktop being powered via the mains always). The three modes are Sleep, Hibernation, and Safe Sleep, and they each work slightly differently. Sleep: This is the default setup for desktop machines. The machine’s RAM is left powered on while it’s sleeping, allowing the Mac to spring into life quickly, because there’s no need to load anything from the hard drive. Perfectly suited to a desktop machine. Hibernation: In this mode, the contents of RAM are copied to your hard drive before the Mac enters sleep. Once the Mac is sleeping, power is removed from the RAM. When you wake the Mac up, the hard drive writes the saved information back to RAM, so wake time is a bit slower. This is the default sleep AT 62

mode for laptops manufactured before 2005. Safe Sleep: The RAM contents are copied to the hard drive before the Mac sleeps, but the RAM remains powered – in the case of a MacBook, MacBook Pro or MacBook Air, that power is derived from the battery if the unit isn’t plugged into the mains power supply. The wake time is pretty quick because the RAM contains the necessary info. However, the RAM’s contents are still copied to the hard drive as a safeguard. Should something bad happen, such as a battery failure, you can still recover your data. The default sleep mode for laptops has been Safe Sleep since around 2005, although some earlier laptops also support Safe Sleep mode. Now if, like me, you leave your laptop sitting around for weeks without being connected to the mains power, you run the risk of draining your battery to zero power. And, as I’ve mentioned, this can kill a battery completely. Ideally the laptop would be set to Hibernation mode to avoid this. It’s a simple process to change the sleep mode with a quick command typed into the Terminal application found in the Utilities folder. To find out which mode your machine is set to, open the Terminal app and type: pmset -g | grep hibernatemode. You’ll get one of the following responses: hibernatemode 0, hibernatemode 1, or hibernatemode 3. For my machine I wanted it set to ‘Hibernation’. To change this setting, type this command into a Terminal window: sudo pmset -a hibernatemode x - replacing x with either a 1, 2, or 3. In this case I used sudo pmset -a hibernatemode 2. You’ll be asked for your account password and then the job’s done – no more RAM draining your battery to death. Now, about that cheap battery. While the battery does work, the actual battery housing was ever so slightly larger than a standard Apple battery, and after forcing it into its nest, the battery pushed against the underside of the trackpad and trackpad button and rendered the button in a perpetual state of pushed downedness – yeah, a right pain. So my next project is to remove the battery cells from the cheap battery and put them into the original battery casing. I’ve got them in pieces now. If it fails, I suppose I’ll end up stumping up for the real deal from Apple. I’ll let you know. Brad Watts has resigned from his role as AT’s Online Editor but will continue to contribute Mac Audio and equipment reviews – as he has done since Issue 1! Thanks for all the hard work while on staff Brad, and good luck. – Ed.

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In a market experiencing upgrade fatigue, Avid hopes the version 10 attractions will be game changers. Text: Brent Heber

PT10 & PCI HD HARDWARE Andy Stewart’s Ed Space last issue raised a few points and questions amongst readers. Principally:: Q: I have an older PCI ProTools HD system, does this work with ProTools 10 on a new computer? A: Yes. If you have PCIx ProTools HD cards, you have a couple of options available to you: 1. Upgrade to a Magma PCIx to PCIe chassis and keep running PCI cards with a new PCIe Mac. If you have a number of PCIx cards this may be the most affordable option. 2. Upgrade your ProTools HD system to ProTools HDX(from $6299). 3. Purchase a HD Native system (from $3883). – Ed.

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It seems only yesterday that Digidesign was harangued for how long its developers were taking to bring an OSX-compatible version of ProTools to market. A significant time lapse had developed between version 5.1 and 6 and all hell was breaking loose, with forums in a foment: “how dare they make me wait this long” etc. Some years later and the release of ProTools version 10 has had the ‘Tools community up in arms again, declaring the very opposite – “I only just bought PT9!” “What are they thinking releasing this so soon?!” Truth is, in the move to v6, Apple owners couldn’t wait to jump onto OSX and all its multifarious attractions. These days, no one’s doing cartwheels to put their heads into the mouth of (OSX) Lion. I think we’re all a little fatigued by upgrade costs, OS changes and downtime with incompatibility issues. In other words, the prevailing sentiment is… please, not another big upgrade. The cost of this upgrade has certainly been a big point of contention. Avid has released the PT10HD upgrade at prices never seen before – the upgrade scale runs as high as US$2500 for a move from PT7HD to PT10HD, down to $999 for PT9HD owners. Historically, an HD upgrade has been around the US$350 mark, so clearly Avid was

bracing itself for an inevitable backlash. And they got one. In response, the Avid Audio division recently extended an olive branch (onto the fire?) by offering the recently upgraded PT9 owners a discounted US$599 upgrade path via an ‘Avid Vantage’ support contract. So the (metaphorical) $64,000 upgrade question is: “Is it worth it?”. The answer is ‘yes’, although not a totally unqualified ‘yes’. Regardless, let’s explore the big ticket additions and you can make up your own mind. These include: Clip gain, real-time fades, 32-bit floating point and interleaved file support, Audiosuite improvements, a Disk Cache feature and an improved plug-in architecture for 64-bit compliance down the track – a handful of these new AAX plug-ins are included with the upgrade. DISK CACHE

The two most important features in that aforementioned list are real-time fades and disk cache. By removing the need to render fades as new audio files, opening and closing sessions is much faster and writing a fade across a huge multitrack session is near instantaneous. ProTools has always used a RAM buffer to read audio files off your hard drive – it’s a setting in your Playback Engine window.

With ProTools 10 that segment of code now runs 64-bit (although the application is still 32-bit) as it can access almost all your installed RAM to store the audio on your timeline. I say ‘almost’ as the software will need 3GB to run, so on an HD-licensed system with 12GB of RAM you can set your disk cache to a maximum of 9GB. If you do, then up to 9GB of audio from your Edit window and Region bin (now called ‘Clip Bin’) will be stored in RAM and played back from there as if from a solid state drive, with, again, near instantaneous playback regardless of edit density and track depth. This feature was developed so PT10 could cope with up to 768 tracks in a maxed out (soon to be released) three-card HDX system, and also means ProTools will play off almost any network volume (I recorded eight tracks to a network share over wi-fi!).

Top Left: With an HD licensed system, Clip gain can be freely interchanged with your volume automation Bottom Left: Session Setup window with the ability to change bit depths and interleaved status on the fly Right: PT10 also features new signal flow interrogation modes previously only available with an ICON


While overhauling the way ProTools relates to hard drives and fades Avid has also reapproached the file types that ProTools natively supports. PT10 will allow files of mixed bit depths to co-exist and playback from the timeline without conversion, and will also support playback of interleaved files without splitting them into their mono channels. This drastically speeds up import times – simply update the wave cache (the file that stores the graphical overview of the waveform) and you’re good to go. Consequently, whether you’re a sound designer bringing in SFX or a musician loading bundles of loops and beats, each time you drag a file in you’ll get to playback much faster. 32-bit audio files and the RF64 file format is also supported, allowing for greater headroom and much longer record times (the RF64 wave file does away with the 4GB file limit imposed in earlier versions). CLIP GAIN

Clip Gain isn’t a DAW revelation – Nuendo has had similar functionality for some years now. At its essence, you get ±36dB gain adjustment in real time on the audio clip (a ‘Region’ now = ‘Clip’) which is both very fast to visually adjust for quick ’n’ easy gain matching and has the benefit of being independent of the volume automation graph (although the two are related). Any clip gain processing is done pre-insert, whereas volume graphing is post-insert. The only way to achieve pre-insert gain adjustments in the past was to insert a trim or time adjuster plug-in and automate it, which could lead to automation lag on playback and was generally a cumbersome approach. Sample-accurate clip gain can be nudged from the keyboard with modifiers and the up/down arrows, and you can also show a clip gain line for flexible adjustment mid-clip. This separation of clip gain from the volume playlist also helps mixers who will now start

audio will be stored in RAM and played back from there as if from a solid state drive with near instantaneous playback regardless of edit density and track depth

RUNNING THE PROTOOLS HD NATIVE CARD Coinciding with my studio upgrade to PT10HD I also took delivery of a new HD Native card. I’ve been sceptical of all this hubbub about an HD Native card outgunning an HD system. I figured most comparisons were probably TDM usage vs RTAS usage, as most mixers are hesitant to combine RTAS and TDM plug-ins in their sessions for fear of breaking the 4000-sample Automatic Delay Compensation limit. This overlooks the advantage of a TDM system having both all the RTAS power at its disposal alongside the TDM chips if you order your plug-ins RTAS first, TDM second. So I built up a comprehensive session, maxing out TDM chips, voice count, bus count as well as RTAS plug-in meters. I did this on a playback buffer of 512 – fine for mixing, not for tracking. The system was an eight-core 2009 Mac Pro with 10GB of RAM. On switching out the HD3 Accel for the Native card and opening the session, the first thing I noticed was how darn quick it was to open up. I expected half the plug-ins to be inactive due to insufficient resources, yet lo and behold, it was all there. Opening and closing this session on my HD3 took nigh on eight minutes a time to program and balance the DSP, on the Native system it took less than eight seconds! In PT10, the native system outguns the ageing TDM architecture in a number of ways I wanted to exploit. First I switched the delay compensation engine up to 16,000 samples (4x the TDM limit!). I also had a spare 50 voices to play with, as a Native 10HD rig can handle 256 voices over the TDM maximum of 192 (at 48kHz). At this point I

hit Play and got an ‘insufficient power’ popup and had to go to my playback engine settings. So I upped my CPUs from 10 to 12 and dropped my buffer from 512 to 1024 and gave it another go. Play. Stable. Quick. Responsive. Wow! So for the difference of my playback buffer and a minor CPU tweak, the HD Native rig absorbed the mixer tasks of the HD3 Accel system into its new 64-bit floating point mixer! Same CPU and yet apparently more RTAS power? The only explanation is that the new 64-bit floating point mixer is a significant step up in CPU efficiency. The plug-ins I used to max out the system were a combination of EQ3 RTAS and the new Channelstrip (which is AAX native), so another contributing factor may well be that the AAX plugs were running better on the HD Native mixer than inside the TDM engine. So if you are running an ageing PT7HD system with two or three cards and thinking of shelling out US$2500 for your PT10HD upgrade, maybe have a look at an HD Native card for US$3500. It will come with 10HD, a new iLok and certainly seems to give a TDM HD rig a run for its money in a modern Mac Pro. Stay tuned for a further report on recording latencies with the HD Native card.

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have been floating your boat so much but a true 64-bit application with access to greater than 4GB of RAM will make a huge difference. The very fact you can load all your samples into affordable RAM is a big improvement and AAX as a format will allow this sort of development to occur. But for now we have the transition to deal with… ProTools 10 allows for RTAS, TDM and AAX plug-ins to all coexist, but Avid has warned that the next major release will drop support for RTAS and TDM. Clearly, there’s going to be some financial pain here, as third-party plug-in developers move in for the upgrade ‘kill’. CHANNELSTRIP

Included in this release are the first few AAX plug-ins: Mod-delay III, Downmixer and Channelstrip. ModDelay is an update of an established Digirack plug-in, but Downmixer and Channelstrip are new additions. Downmixer allows editors in stereo rooms to monitor sessions that are routed in 5.1 or even 7.1. Previously this would have been a third-party plug-in bundled with Complete Production Toolkit, now it’s included with an HD license as standard.

Avid Channelstrip showing its Average/Peak detection modes and the selectable gain structure.

with a fresh clean volume line, rather than dealing with editors’ squiggles! Region groups have been updated to ‘Clip Groups’ and now gain levels will remain with the groups as they move from ProTools session to session.

NEED TO KNOW Price ProTools 10: $739; ProTools 9 upgrade: $319 ProTools LE/MP crossgrade to PT10: $529 HD users: Priority Support includes PT 10 and software upgrades for 1 year $665 Contact Avid Pros Much faster operation generally Co-install RTAS/TDM and AAX Starting to use 64-bit technology Useful new plug-ins included Cons Cost of upgrade to 10 and upgrading plug-ins to AAX Still negligible improvement to instrument performance Still no ‘faster-than-realtime’ bounce Still no ‘Freeze tracks’ function Summary A great step forwards for users of ProTools in TV and film. A more incremental step for composers and music producers.

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ProTools 10 files will now bear a PTX extension, as opposed to earlier PTF files. This is to accommodate the new fade and file structures, however, it means to collaborate with studios on earlier releases you will need to do a “Save Copy In” and lose your separate clip gain vs volume automation. Some niggles well inevitably result. AUDIOSWEET

Another big area that has been overhauled is the Audiosuite/rendered processing. You can now open multiple Audiosuite windows at once and if you select ‘Clip by Clip’ and ‘Individual Files’ you can retain your fades and handles when rendering effects. If you wish, you can even opt to apply an Audiosuite process to a whole file, even when only a snippet is visible on the timeline – so you never have problems un-trimming again. Windows configurations can be used to store multiple Audiosuite windows being open at once and will also store the settings of the plug-ins and recall them at the same time. This is potentially a massive timesaver for folks who need to render effects in order to pass projects around between studios/rooms. MIX ADVANTAGE

Up to this point, you can see that keyboard and mouse editors will get solid improvements in how they approach their work in 10 – what about the mixers out there? The biggest deal on that front will be the adoption of a new plug-in format: Avid Audio eXchange or AAX. The key move here is in making the ProTools proprietary plug-in world 64-bit to access more RAM. And here’s where the composers amongst you should be sitting up and paying attention – the other stuff may not

Channelstrip is quite an addition, containing the DNA of the EQ and dynamics section of Euphonix’s flagship System 5 mixing console. I say ‘DNA’ because it’s not so much an emulation as it is a stem cell ‘clone’. There’s input and output trims, some versatile filters, a solid EQ and a very interesting dynamics section. Any of the four components can be bypassed from the FX Chain view at the top of the plug-in and clicking on the context menu allows you to reconfigure the signal flow within the channel, exactly as you would on a System 5. As someone who works on an Icon mixer I have a few gripes with how ChannelStrip maps onto the Icon control surface, but when it comes to the sound I have no complaints. The compressor is extremely transparent on first use, but then you realise it’s because of its attack time behaviour – if you open the Sidechain page you can switch between Average compression and Peak compression, and by default it’s set to Average, unlike most compressors. Hence, it lets through many peaks which gives a lot of air to the resultant signal. Also, don’t expect this compressor to act as a brickwall limiter, as it doesn’t have the ability to look ahead “because the System 5 console doesn’t work like that”. In my opinion I think this is taking the ‘faithful recreation’ approach too far. If I was Avid I’d want ProTools users (accustomed to look-ahead ability) with no experience of the System 5 to be blown away in every respect. Instead, I’ve found myself reaching for other limiters alongside Channelstrip as its fastest attack speed seems to max out at 15ms. THE SHAKEDOWN

It’s much faster on big sessions in all the ways that have frustrated me in the past and I appreciate the increased trackcount, better workflows and new plug-ins. This release definitely has more of a post-production focus compared to the more music-centric release of ProTools 8 (with all its MIDI functionality). It’s also a relief to see Avid moving towards 64 bit (albeit slowly) with its AAX plug-in format but is this a ground-breaking, innovative release? Not really. It’s more an evolution of a system based on customer feedback and competitive pressure over many years. That said, it’s certainly nice to see they listen to our feedback even if it takes a bit of time to get to market! So how does it all shake down? I’m happy as a pig in muck on ProTools 10.








The Maschine world has expanded in three major ways: Maschine Mikro is the perfect entry to pro beatmaking. With its compact, go-anywhere controller and same powerful software as its big brother, Mikro is unbelievable value. And now, both Mikro and the flagship Maschine receive a brand-new 1.7 software update, featuring seamless integration with Komplete 8. And if that’s not enough, check out iMaschine, the mobile beat-making app for your iPhone®, iPad® or iPod Touch®–available from the App Store**. Whatever your workflow, there’s now a Maschine version for you.

Distributed by CMI Music & Audio: Apple, iPad, iPhone, iPod touch, and iTunes are trademarks of Apple Inc., registered in the U.S. and other countries. App Store is a service mark of Apple Inc. *Based on current street pricing. Consult your NI Maschine dealer. **Visit for system requirements. AT 67



Can this self-contained rackmount mini-monster survive a Bluegrass acid test? Text: Mark Woods

NEED TO KNOW Price $22,780 RRP Special pricing: $9990 (until January 23, 2012.) Contact Group Technologies (03) 9354 9133 Features Desktop or rack-mountable 16 local mics pres, 8 outs D-Rack expansion adds another 32 ins/16 outs 2 x AES/EBU I/O MIDI, MADI port, GPI/O 3 x USB ports 8 x FlexiChannels/12 x FlexiBusses Pros Great sound quality Control options Flexible operation Very compact Cons Screen in daylight Noisy fan Summary A complete high-quality mixing/processing system in an efficient and portable package suitable for a wide range of touring or installation applications. AT 68

Regular readers may recall me raving on about how much I enjoy mixing at the annual Harrietville Bluegrass Convention, and this year’s event was no exception. Beautiful music in a great mountain setting, fantastic players, southern American accents and an appreciative audience. And it goes for four days so it’s a good gig for trying out new gear. I always seem to have a couple of new mics to try but the big one this year was Digico’s newest and smallest console, the SD11. It arrived literally a few hours before I hit the road so all I had time to do was re-route the channels to receive/send audio through the local inputs and outputs on the rear of the console, and throw it in the car (not literally).

use an SD9 about a year ago and the SD11 arrived without a manual or any instructions other than the on-board menus. I’d considered using the Digico for the main stage but for this event, more than most, I wanted to be able to react quickly and confidently to situations that arose during the main concert performances, and considering how long I was going to be mixing I didn’t want to have to think too much… so I chose to use an analogue board for the main stage at Harrietville and the SD11 for the B stage. Moving between the two consoles several times a day turned the event into an ongoing A/B comparison between the two technologies that clearly defined the differences, both physical and mental.



The SD11 is a small-format digital console that can be rackmounted or used on a desktop. It packs 12 motorised faders, buttons and knobs for selecting and adjusting parameters, a touchscreen and a comprehensive list of I/O options: 16 XLR inputs and eight XLR outputs on the rear panel make it a compact but fully-featured stand-alone mixer. Connection to the optional D-Rack via Cat5e cable adds another 32 XLR ins and 16 XLR outs. The screen dominates the operating surface while the knobs and buttons are large, well-labelled and easy to reach. The touch-sensitive faders feel good in the hand. Overall it’s a great size and quite easy for one person to carry. It’s not exactly light at 24kg but feels very strong. I’ve used quite a few digital consoles over the last 10 years and while I haven’t owned one I’ve been able to get familiar with several brands and models for both recording and live use. But I still get insecure when faced with a new one. It’s not the audio quality or controls that worry me, it’s the routing and setup options. Show me any analogue board and I’ll delight in exploring the functions, only occasionally finding something that requires explanation. With the Digico, well, I spent a couple of hours being shown how to

The local I/O on the SD11 made connecting it to the system easy and familiar and it took up no more room than my effects rack sitting next to it – which was made redundant by the SD11 anyway. On firing up the PA my first impression was the excellent sound quality. Digico has a reputation for making the best-sounding digital consoles and deservedly so. The SD11 has the same basic sound technology as the other models in the range, and I was very happy with it. The high frequencies are the hardest for digital equipment and some of the more popular digital consoles I’ve used sound harsh and/or grainy. The SD11 is not a bright-sounding console but it’s super smooth in the top end with a warm and accurate midrange and easy lows. Headroom seemed high and the overall sound had a sense of power that took loud peaks and transients in its stride. Vocals and instruments were pleasingly natural and in a blind test I wouldn’t have known it was a digital console. CONTROLS & PROCESSING

And then there’s the extensive controls and processing. Input channels have delay and moveable insert points. The EQ options are powerful and comprehensive. The standard channel EQ consists of four-band parametric controls plus

HPF and LPF. I generally prefer digital EQ to analogue and the SD11 does what you want without any artefacts or weirdness, even with large amounts of EQ. Twelve virtual 32-band graphic EQs can also be summoned and you’ve gotta love the way the faders become cut/boost controllers for each EQ band, complete with centre indents. The standard channel EQ can be accessed quickly by touching the EQ section on the virtual channel strips on the screen or, if you have selected the appropriate channel by touching the strips or the fader, by adjusting the EQ knobs to the right of the screen. Not as fast as analogue but close. DYNAMIC PERFORMER

For mine one of the best features of digital consoles is the availability of dynamic controllers and here the SD11 excels. Each channel has a gate with key function and a single band compressor. This can be turned into a three-way multi-band compressor at the touch of a virtual button and up to four instances are currently available. The turnover frequencies can be adjusted and there’s Auto-Gain and Listen on each band. Another great feature is the way separate dynamic control can be applied to each of the four EQ bands – again up to four instances. You don’t get that on analogue boards. All these dynamic controllers work smoothly and predictably, and like only the best analogue compressors they don’t dull or distort the signal, even with high levels of gain reduction. Input level and gain reduction meters on each virtual channelstrip help to monitor proceedings. This quantity and quality of dynamic processors would require a vast amount of hardware and cabling in the analogue world and digital wins this one hands down. EFFECTS ENGINE

The other big saving is the built-in effects. There are four effects engines providing a comprehensive range of the most common effects and they’re good quality. I had some trouble getting them patched but figured it out eventually. With audio gear, part of me thinks that if you know what you want to do, and can’t figure out how to get it working easily then it’s bad design, but I can also accept that these are complicated, multi-function machines and a little qualified instruction may be required at first. Effects are also available via the optional SoundGrid module that links to an external PC to enable the use of up 16 Waves processor racks, or you can register and run TDM plugs too. The channels are largely user-definable and the four banks of 12 faders and four layers control up to 32 channels, eight of them in stereo. 12 FlexiBusses are available and can be configured as mono or stereo. Eight Control Groups can be used as VCAs, moving faders or mute groups. Output options include L/R, LCR, and the 12 FlexiBusses can be configured as mono or stereo Groups or Auxes, in addition to an 8 x 8 matrix. There are two AES/EBU in/outs and a MADI interface allowing up to 56 channels to be recorded on an external device. FANCY PICKING

At Harrietville the B stage is the more demanding environment: it’s a big marquee that has a stage at one end, a bar at the other and a food area in the middle. Its noisy most of the time and it’s a difficult situation for acoustic music that tends to get swamped by background noise. Couple this with condenser mics on stage and the enthusiastic organiser’s desire to have it as loud as possible and it turns into a gig where every dB counts. To get the most from the PA required not only a steady hand on the faders and some fine control of channel/overall EQ but a high level of dynamic control to deal with the wide range in input levels. In this situation the SD11 was fantastic and

after a short while I’d built up a mix that was stable, loud enough and didn’t freak out when performers moved from playing a meter away from the centre mic to a few inches for their instrumental breaks or vocals. It attracted some attention too, with a steady stream of interested punters and musicians asking me what it was and how it works. I take flying faders for granted now but they still impress certain punters. It’s not all sunshine and light though. The biggest problem I have with all digital consoles is the inability to see the touchscreen in bright light… like daylight. Even in the shade it is very difficult. I’ve tried shading it with my hands or using shades but the damn things are just not bright enough. There is the option of plugging in a computer screen and they’re a little easier to see… but, of course, a PC monitor isn’t a touchscreen. I believe the problem is with touchscreens rather than the console manufacturers but it’s annoying and I’m sure I’m not the only one who sometimes does shows in the daytime. Screen developers please note: there’s money to be made with a touchscreen you can see in bright ambient light. While I’m trying to find negatives I’ll mention the cooling fan: it’s noisy. It’s noticeable at live shows and gets hidden easily come showtime but it was a problem when I was checking the desk out in my control room. The SD11 is not really designed as a studio console but it’s so portable I can imagine some users using it for both situations. [Digico has now released the RR-PSU to allow remote location of the power supplies when audiences are particularly fan-noise sensitive – CH.] TURNED UP TO 11?

Most of the shows I do now are one-off or annual festivaltype events; they need gear that is simple and fast to operate and analogue consoles work well for me, although I can see that situation changing if I owned a digital console and could get super-familiar with it. When I was touring regularly I liked to build up specific settings for channels and effects for each song. Digital consoles excel in this area and if I was doing tours again I would definitely be using consoles like the SD11, or others in the Digico range, and I can understand why they are popular in the high-end touring market. The Digico SD11 is on the upper end of what you’d expect to pay for a console of this size/ functionality/channel count but when you factor in its heritage, sound quality and robust construction the price looks better. It’s tempting to think of digital consoles as simply computers with converters and faders, and I’m sure they will get cheaper and cheaper as time goes by, but this is a quality product that is software updateable and should endure for some time to come.

SD11i SOFTWARE UPGRADE Digico has announced a new software package called SD11i – a free upgrade that expands the SD11’s capabilities. The input channel count goes to 32 full FlexiChannels, giving 64 channels of DSP. The multiband compressor, digital FX, and dynamic EQ all increase from four to six. Additionally, six channels of DigiTube are now available which is Digico’s valve emulation previously only available on the SD7. Further new features include an expanded Aux Panel, increased dynamic functions such as a duck option and expander for the gate, a de-esser and two types of knee for the compressor. There are also new FX types with multitap, ping pong and stereo delay, with different delay times for left and right. A Warmth button is provided on each channel for analogue emulation and two types of EQ have been added (Classic and Precision). There are also now Standard template sessions. Another new feature, Sets (previously only available on SD7T), has been enhanced to allow for ‘Set Spill’. This allows the user to create Sets and, with a simple button press, change the console layout to display members of that Set. AT 69


PROPELLERHEAD REASON 6 & BALANCE AUDIO INTERFACE Is the sky falling in? No – Propellerhead’s Reason virtual electro studio has finally grown audio tracks. Text: Derek Johnson

At one time, the idea of audio recording being added to Reason – the king of virtual electronic music studios from Swedish coding chefs, Propellerhead – seemed highly unlikely. I don’t think the words “It will never happen” were ever stated, but one never felt encouraged that it would. The release of Record, a wonderful-in-itsown-way multitrack audio recording package, may have give cause for some optimism, but Reason+Record Duo seemed to be the way of the future: choose electronic music studio, choose audio recording, or choose both as an integrated bundle. But two pieces of software, even as wellintegrated as these two were, was still less convenient than one, not to mention the extra load on the host computer and the wallet. Distributors, dealers and some users found the situation a bit complicated so, after a couple of years trying to satisfy two different types of user, the Swedes made the integration permanent. Record has been quietly phased out, with all of its functionality – and then some – inherited by Reason 6. The new package even costs rather less than the old Reason+Record bundle. To coincide with Reason 6’s release, Propellerhead has taken its first step into the AT 70

hardware market, with the Balance two-in/ two-out USB2 audio interface. The market Propellerhead hoped to build for Record continues to be serviced by Reason Essentials, a cut-down (rather than entry-level) package aimed at bands and songwriters. This version 6 release also sees Reason finally working on 64-bit platforms, Mac or PC, if your host of choice is right up to date. REASON TO BE CHEERFUL

Summarising Reason 6 as it now stands would bust my wordcount, so I suggest you trawl the AT online archives, and fill in any gaps with Google. For the completely uninitiated, let’s just say that Reason started out as the virtual representation of the most flexible electronic music studio you could never afford. It’s based around a collection of well-designed analogue-style and purely digital sound generators, sample and loop players, and effects. The synth devices include a mix and match of classic analogue and digital synth modelling, with the mighty Thor easily capable of standing alone as the source of an entire track. Sample players are also available in simple and overkill varieties: NN19 for the simple jobs, and the jaw-dropping NN-XT for mega multisamples. Among the drum

REASON ESSENTIALS Reason Essentials takes the Reason 6 design and pares it right back to the basics. The result is a perfect, easy-to-use, environment for bands, singer/songwriters and anyone who might prefer to not be distracted by the full weight of everything that Reason 6 offers. In fact, it feels rather like a nod at the market that Record was initially capturing. Essentials features unlimited audio tracks, plus timestretching (but not transpose), a basic mixer, and a limited array of sound-making and effects devices. The Line 6 Pod bass and guitar amps are here, along with the four MClass mastering effects (Compressor, Limiter, EQ and Stereo Imager), plus RV7000 reverb, a delay, a chorus/flanger, and the Scream 4 Distortion. Sound-wise, you’re gifted the ID8 sound module, Subtractor synth, Redrum drum machine, NN-XT sampler, and Dr Octo Rex multiple REX loop player. There’s a few other bits and pieces, backed up by a smaller factory sound band than Reason 6. For getting ideas fleshed out quickly, Essentials offers a great environment and it’s possible to mix and polish to a high standard (I’m a big fan of the MCLass suite). And you can always upgrade to Reason 6 later, of course.

machines, ReDrum feels very ‘Roland beatbox’, and the Kong Drum Designer would be an MPC killer in hardware form – it features sound generators, sample and loop players, and you can create kits of great depth and sonic sophistication. If you’ve missed a Reason upgrade, Dr Rex no longer plays REX loops – but multiply that by eight, and you have the compatible but so much better Dr Octo Rex, from Reason 5. Add a step sequencer, arpeggiator, a couple of mixers and tons of effects devices including the luscious RV7000 reverb and MClass Mastering collection. Some devices are simple, some are so sophisticated and flexible that you might wish they were sold separately. You, the user, load the devices into an on-screen rack (now double width in Reason 6), link them with virtual control and audio cables, and create music. It starts out simple but can become as complex as you like – the linking is largely unlimited, and some deceptively simple devices let you mix and split audio and control signals alike to set up massive patches, effects groupings and so on. The dedicated Combinator, introduced a few upgrades ago, makes multi-synth/multi-effects groupings behave as one device-like patch, tidying up the rack. Reason offers a flexible sequencing environment that lets you work with patterns, blocks, or completely linear tracks. The feel is old-school but all the better for it – recording and arranging are streamlined here, with full automation of nearly every sequencer (and rack) function. There’s still no MIDI Out, but Reason 6 has tight integration with external hardware controllers so that you’ll be able to plug in virtually any keyboard or control surface on the market and have it wiggle some of the plentiful on-screen controls in a useful manner. IN WITH THE NEW

New for Reason 6, and inherited from Record, is a global mixer ‘modelled’ on SSL’s 9000K behemoth. It grows a new input whenever you create a new device or audio track; in fact, an audio track gets its own device-like entry in the rack, so you can easily integrate audio with whatever else you’re doing within the rack. That means you’re working

in essentially three modes – Rack, Sequencer and Mixer, though the sequencer has two sub-modes, dubbed Song and Edit. Actually, there is also a Block mode in the sequencer (new in Reason 5), but it’s really another way of editing, extending or mapping out your song. If you needed an excuse to work with three monitors on your desktop, Reason 6 could be it! In another context, the mixer would have its own review. It has built-in dynamics processing and master compressor, room for inserts aplenty on the channels and in the master section, four-band EQ, eight pre/post effects sends, metering and level/pan control. All mixer facilities can be automated and integrated into the rack environment. About all that you’re missing here is subgrouping and surround mixing. The new audio tracks are accommodated in the same way as any other device: add a track, and it’s added to the sequencer track list, an Audio Track device is added to the rack, and the mixer gains a channel. As with Record, capturing audio performances is a doddle. The processing needed to time stretch and pitch shift audio is added invisibly in the background; you can turn these options off if you require un-stretched audio. The process, by the way, is stunning, even up to more than modest stretches. EFFECTIVE CHANGES

There are no new sound-making devices as such, though the ID-8 sound module, with its preset bank of useful patches, has moved over from Record. The factory sound bank has been hugely expanded, too. There have been additions to the effects arsenal, though. Line 6 guitar and bass amp devices have come over from Record, adding a unique slice of non-Propellerhead design to the Reason environment. Also from Record is the wondrous Neptune pitch adjuster and voice synth (fix bad notes/add harmonies/wig out). Now to the new toys. The Echo is clearly modelled on classic tape delay effects, especially those produced by Roland. We get warm, fuzzy sounds here but with a 21st-

Reason 6 now has three major operational windows plus a number of smaller ‘navigator’ areas to help with moving around busy songs. Each one of the main windows can be given focus, or split off to a second monitor, to allow for easier operation.Pulveriser adds dirt, depth and excitement to input signals. Bringing the concept of tape echo into the 21st Century, The Echo offers complete control over every aspect of the stereo delay process. Capable of practically creating a new sequence or pattern from any input signal, the rhythmic dynamics and filtered madness of Alligator cry out for extended tinkering.

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CLIP SAFE One of the best things about the Balance/Reason combination is Clip Safe, enabled by pressing a special button on the interface. The example shows some newly-recorded audio in Song Edit Mode. Clipped audio is indicated by the red lines in the strip above the waveform display; this is an exaggerated example, and has a chunk of waveform with a that’s flat-lined to the left. See the little ‘CS’ button to the top right of the upper display. Click it and the result is picture two – healed audio. Not only have the red lines gone, but so has the plateau in the left of the waveform, restoring exactly what I’d meant to record in the first place.

century edge that makes you wonder how you managed without. it. Plenty of feedback, modulation and external control options will keep this one ping-ponging in all my future racks.

NEED TO KNOW Price Reason 6 $499 Reason Essentials 1.0 $279 Balance with Reason Essentials $499 Contact Musiclink Australia (03) 9765 6530 Pros Audio recording – with timestretching. Three new desirable effects. 64-bit CPU support.

The effects produced by Alligator, a three-band filtered gate device, could conceivably have been created with multiple devices in the past, but it would have been complicated. Alligator splits a signal into three channels, with each channel treated with its own patter-based rhythmic processing, filtering, modulation and effects – distortion, phaser and delay. The highs, mids and lows of the input can all be treated differently. Has to be heard to be believed. Reason already has a capable distortion device, in Scream 4, but the new Pulveriser ‘high yield demolition’ module goes into different provinces of carnage. This combination of on-the-edge compression, distortion, multimode filtering and modulation somehow manages to mix digital and analogue power and destruction without harshness (unless you want harshness). You can really squash a sound flat or bring out its inner detail in unexpected ways. ALL THE RIGHT REASON

Cons Still no third-party plug-in support. No surround mixing.

There really are few downsides to Reason 6. Sure, some of us don’t like dongle-based copy protection, but we don’t like piracy either and dongles do help. The absence of surround mixing may bug some users. The ongoing lack of third-party plug-in support will certainly bug many users, but what you can’t do in Reason... well, just do it in another package, or ReWire Reason to another DAW.

Summary Propellerhead keeps the ball rolling with meaty developments to its core software – and the longwished-for audio recording. And we have hints of a whole new game following its release of a top-notch but affordable hardware interface.

For all-in-one electronic music creation, especially but not exclusively that which focusses on looped audio or patterns, there’s little to beat the speed and power of creation in Reason 6. The audio tracks are flexible enough to allow full band recording, and integrating Reason device patching with a guitar/bass/drums/vocals outfit takes each medium to different places. This isn’t a unique idea but Reason just makes the process so much fun. So, existing users should have already upgraded – even those who disdained Record. If you’re new to the market, just go straight to the top. You won’t regret it.

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REASON ESSENTIALS WITH BALANCE Bizarrely, Propellerhead’s first foray into hardware design – two ins and two outs of USB2 powered audio interfacing – is being packaged as an add-on in a bundle with Essentials! As it happens, Essentials is a great companion to the interface, and the bundle makes a perfect songwriting tool for the desktop muso. Balance is an almost perfect device, marrying features, sound quality and elegant design in one fab package. Even its finish is attractive. A copy-protection dongle is also hidden inside Balance, which is quite an elegant touch. It’s a 24-bit device, handling sampling rates up to 96k. In terms of inputs, you’re offered a left/right pair of independently switchable phantom-powered XLR mic ins, two pairs of balanced line ins, and two guitar level ins (each with pad). Switching buttons on the top of the box let you select one each of any pair at any time so you can record a vocal while thrashing an electric guitar, for example. There is a pair of input level controls, a load of signal and overload LEDs, and big knobs controlling main (balanced line) output and headphone levels. The headphone socket is not of the mini variety: it’s the full quarter inch. Other features include a record ready indicator for each channel, which is specific to use with Reason. The same goes for the Meter/ Tuner button (it calls up an on-screen meter or guitar tuner). Then there’s the Clip Safe button, which evokes a simple but fantastically useful feature. With a mono input, enabling Clip Safe causes two recordings to be made, with one using more headroom. So, if you should record the perfect take but have accidentally clipped the audio in a couple of places, you don’t have to redo the performance. Pressing an on-screen button ‘heals’ the audio by replacing the clipped sections with clean audio, and gives you a pristine take. Beautiful. I hoped that Balance would have had digital I/O on board, and the lack of a power source other than the USB connection also causes concern (it is being asked to fire up two phantom-powered mic inputs, remember). But there were no problems in testing, and if a laptop user needed a wall wart, she’d be able to plug in her laptop anyway. Bottom line is that I recommend you give it a listen if you’re in the market for a high-spec desktop audio interface.

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STEINBERG CMC CONTROLLERS We take a sneak peak at Steinberg’s modular approach to Cubase/Nuendo control. Text: Graeme Hague

Ah, controllers… they say that once you’ve used ‘em, you’ll never go back to the archaic RSI-inducing method of working with a mouse and trying to remember the myriad key controls assigned to your computer keypad. I must be one of the exceptions, because while I reckon that a good controller can look impressive on your desk and a set of nice faders definitely promotes a hands-on mix, my old habits of mousing the finer parameters of a DAW refuse to die. I don’t have to think twice about what I’m doing – like, whether I’m going to grab the right knob on a generic controller. And you have to agree that mouse technology has come a long way with ergonomic shapes moulded to your hand, responsiveness that can be fine-tuned and even selectable weights. Do controllers still really offer a significant advantage? Steinberg obviously thinks so, but it has come up with a different approach to controller hardware for its Cubase and Nuendo DAWs. For a start, the CMC devices – I guess it stands for Cubase Modular Controller – are certainly made by Cubase for Cubase with next to no allowances for them to work with other software. All the button icons and much of the physical layout of the controls directly mimic what you’d get on-screen in Cubase, so there’s no doubting or second-guessing the purpose of each button and slider – for Steinberg aficionados anyway. It’s not an entirely new idea, but in the cut-throat market of DAW applications and peripheral gear it’s a bold policy – you’re a ProTools, Sonar or Logic person? Bad luck. PICK N MIX NEED TO KNOW Contact Yamaha Music Australia 1800 331 130 steinberg_australia@gmx.

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Steinberg has designed six separate devices, each practical in their own right depending on your individual needs, but at the same time various combinations won’t needlessly double-up on functions. You can also have more than one of each CMC, if you like. So two of the CMC-FD fader controllers will provide eight channel faders. You need 16? So buy four CMC-FDs. Hell, buy a couple of dozen and you might be able to mix down the

guitar tracks on a Brian May tune [sorry, Graeme, the limit is four – Ed.]. Steinberg has opted to incorporate what it calls ‘high resolution’ touch surfaces rather than actual faders. It’s allowed for very slim and lightweight designs. However, it immediately makes me question whether these touch surfaces are a contradiction to half the appeal of controller hardware – that tactile, old-school feel of a genuine fader under your fingers still isn’t there. It’s like you’re just swapping one virtual input for another, so what’s the point? CMC MACHINES

To make any of the above CMCs work you need to download and install a required version of Steinberg’s Advanced Integration (AI) software which takes the pain out of Cubase quickly recognising and initialising each CMC as it’s connected and powered via USB (Cubase AI6 is included in each CMC box). All the units have a fold-out leg to angle the face and they can be clipped together to stand side-by-side as one piece. At 92mm wide and 182 mm high they’re hardly the portable, ‘pocket’ device Steinberg claim and, in fact, either by design or coincidence, four of the CMCs are very similar in size to a normal computer keyboard. An optional Studio Frame 4 will house four of the CMCs or the CMC CC121 Extension Frame lets you put one CMC either side of a 121 Controller. My only beef is that there’s no means to daisychain the USB connections. If you choose to buy all six of the CMCs you’re going to need six USB ports. Any decent powered hub will easily cope, but you’ve still got six cables going somewhere. All combined, the six CMCs will make for a reasonably expensive control surface. Individually, apart from the CMC-CD, they certainly do the job of getting your hands off the mouse. Choose carefully or buy the lot, they will make a big difference to your Cubase studio workflow.

CMC-CH: $199.99 The Case: The CMC-CH is a straightforward mirror of the standard Cubase channel strip with the fader represented by one of the high-resolution touch strips. Sixteen buttons and a single rotary button provide channel selection along with all the usual suspects such as MSR, Read and Write for automation, Monitor plus launching inserts, EQ and Sends. An Edit button opens the larger plug-in GUI interfaces and you have an Edit Instrument key. There is also a Folder button – because Cubase has track folders – and a Freeze button. The rotary knob is for Pan. A Shift button changes all the above into Function keys (F4, F5, etc) or turns the Insert, EQ and Sends buttons into a Bypass for the same.

The Verdict: Apart from the fader just about every function here will send you straight back to the mouse pad, so it’s hard to see the CMC-CH achieving much on its own. Still, I can’t imagine Steinberg not creating this module as a kind of core unit to the whole series. The CMC-CH will be at its best in conjunction with one or more of the others hooked up, too.

CMC-FD: $249.99 The Case: The CMC-FD offers four of the newly developed, touchsensitive faders to provide precision control over four Cubase channels simultaneously. I lied about the Brian May thing before – at present you can only connect four of these controllers at a time to make 16 channels. The touch faders are dual-mode, turning into LED-based metering, plus there is a solo/mute function by tapping the fader surface – oh, and you’ve got Channel or Bank selectors. But it’s all about the faders which have an ‘ultra-high’ resolution of up to 1024 steps. I reckon that means my fat fingertip will cover about 100 then.

The Verdict: The biggest and only question is how well the faders work and whether they offer a better, more controlled touch for your mixing and since this is a preview I’m afraid the verdict will remain behind closed doors until we can get our hands on one.

CMC-QC: $199.99 The Case: LEDs that change their intensity according to the set position. The controller comes with three dedicated working modes — EQ, Quick Control and MIDI. In EQ mode, the rotary encoders are the Cubase Channel EQ controls with frequency, gain and bandwidth levels. Quick Control Mode is a handy means to assign any parameter to any controller using a Learn function. MIDI mode turns the CMC QC into a freely configurable MIDI controller. Using the included CMC-QC Editor software you can edit the MIDI control change parameters via a graphical interface. Parameter values are displayed underneath the respective encoders and can be entered manually.

The Verdict: Pretty much anything can go anywhere, so the CMC-QC looks like a good partner to the CMC-CH for a start. There isn’t much, if any, indication of what you’ve assigned to where, in either QC or MIDI modes, so you might be testing your grey cells, but it does bring that all-important tactile interaction with finicky parameters through the rotary encoders.

CMC-PD: $199.99 The Case: The CMC-PD has 16 high-response pads for programming drums, samples and the like into your music. Because it has an MPC-style 4x4 matrix and Cubase’s own Groove Agent One uses the same, there’s a match made in Heaven here... well, in Steinberg Heaven at least. But by using the included PD MIDI Editor you can assign anything to the pads, which makes the CMC-PD a handy little beastie indeed. A rotary button works with a Browse key for scrolling through sample libraries. The four Velocity Mode buttons makes for easier Step Mode recording (if you’ve never used Step recording, that won’t make sense – but don’t worry).

The Verdict: One of the must-have CMC units. You could just about turf your music keyboard in the bin, if you’re not a piano player, but let’s not be too hasty.

CMC-TP: $199.99 The Case: The CMC-TP is a Transport Controller with all of Cubase’s normal transport controls plus a slider that offers (take a deep breath) jog, shuttle, locate, scroll, zoom, metronome and tempo. A shift key adds useful extra functions to the buttons such as Marker insert and placement and Nudge, and there is a Copy Track key – which strikes me as what might have been a “we’ve got one button left, what are we going to do with it?” moment in Steinberg R&D. So they chose Copy Track. Four of the buttons can be freely user-assigned.

The Verdict: As someone who is often self-recording and has to stretch for the Transport buttons while hanging onto a guitar, dodging a microphone and trying to keep cans on my head – you never say no to a well-configured Transport Control that you can plonk within reach. The clever slider thingy gets a big thumbs-up as well. The CMC-TP goes on the must-have list, too.

CMC-AI: $199.99 The Case: The CMC AI Controller is centred around Steinberg’s AI (Advanced Integration) Knob, previously (and still) seen in the CC121, CI2 and CI2+ interfaces. This ‘point and precision’ knob allows you to quickly control any visual Cubase object, such as mixer faders, EQ gain knobs or effect send levels. By pointing the mouse cursor on the chosen parameter or virtual device you instantly control it with the AI Knob without additional actions like clicking – that’s the clever bit. The object can be locked to the AI Knob to retain control when the mouse is moved elsewhere on the screen. There is also a Volume Mode for... well, take a wild guess, and a Jog mode to turn the AI Knob into a proper Jog Controller. Four user-assignable keys are provided here too – always useful.

The Verdict: The AI Knob on its own with its intuitive selection of anything under the mouse cursor makes this a clever piece of gear that will get a lot of use. Yes, Volume Mode is a bit silly and I know that a lot of video editors can be picky about their jog wheels and how well they work, but as it is the CMC-AI is a great idea. I want one.

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LINE6 XD-V WIRELESS SYSTEM Line6 shakes up the wireless market with a system that’s got plenty going for it. Text: Christopher Holder

NEED TO KNOW Price V70 (Handheld version): $699 V70C (Lapel version): $699 V70HS (Headset version): $799 Contact Musiclink Australia (03) 9765 6530 Pros Sounds great. No interference problems. Easy to configure. Choice of mic models very useful. Cons Max of 12 channels. ‘Saggy’ mic clip with weight of transmitter Summary A well priced, well designed, great sounding wireless system that’s easy to learn and configure. Well suited to just about any small to medium application.

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Imagine a vocal mic that allowed you to almost instantly swap capsules: SM58 for one gig, Audix OM5 the next; Beta 58 one song, Sennheiser e835 the next. Or put it this way, have you ever taken one vocal mic to a gig then realised the vocalist was totally unsuited to it and you’d be much better off with another? Sure, you can make it work, but we all know how important it is to find the best mic match for a vocal, and sometimes that search can last years. SEARCH FOR A SUPER MIC MODEL

Line6 does a wireless range called the XD-V and the handheld mic in the V70 system allows you to do just that. There are six virtual mic models onboard, based on the performances of Shure’s SM58, Beta 58, the Sennheiser e835, Audix OM5, Electro-Voice ND767a, and the AudioTechnica AE4100. With a few swift taps on the two buttons on the handheld you can switch between models. Are they electron-for-electron clones of the real things? Who cares. They’re close enough and ultimately the best thing about it is you instantly have more tools in your locker. Only last weekend I had a guest vocalist who I started on the OM5 mic model, only to quickly realise a trusty SM58 was going to be way more suitable for the idiosyncracies of her voice. So between songs during soundcheck I made the virtual switch on the mic. Immediately I felt like I was way more in the ‘ballpark’ than I ever could be with the three-band EQ of my analogue mixing console. You might be thinking: that’s a good party trick, but hardly worth wagering the farm on. And, to be honest, when I first heard about the Line6 wireless I was of the same opinion: “here’s a company with instrument modelling smarts trying to find new ways in which to package its IP”.

Turns out the mic modelling is only one aspect of a really well conceived system. 2.4GHZ: WHAT’S THE DOWNSIDE?

XD-V is a complete wireless range. Handheld, guitar, lapel, headset; there’s a system for most applications. They all share the same RX212 receiver unit. So effectively XD-V is no different to any other wireless system, like a Shure SLX, except in one key respect: XD-V works in the 2.4GHz range, not UHF. As you’re probably aware, 2.4GHz is a popular frequency. Wi-fi and Bluetooth use 2.4GHz, so in other words, unless you’re reading this article in The Kimberley, you’re bathing in it – your cordless phones, wireless router, wireless mouse etc, are all operating on the same frequency range. The reason why 2.4GHz is so popular, is because it doesn’t require a license. And from an audio perspective that’s great news: you have a system that will operate anywhere in the world – no need to worry about finding a slice of available UHF spectrum or experiencing random interference. Brilliant! So why then doesn’t every wireless mic or in-ear system use 2.4GHz then? To operate in the 2.4GHz range you need to work digitally – unlike UHF which works with analogue signals. So, to be ‘heard’ in the crush of this wireless digital blizzard we’re engulfed in, you need some fairly serious error correction – packets of data will be dropped hither and thither so you need to employ methods whereby your receiver takes a look at the data stream from a variety of ‘angles’ and grabs the one with the most integrity [check out the Digital Diversity box item]. All this cross-checking takes time. And this traditionally has been the main killer: latency. Line6 quotes a latency figure of 4ms. This is loads better than

earlier 2.4GHz wireless systems by the likes of Sabine, but many in the pro sphere would contend that the 4ms figure is ‘still too high’. For most small to medium applications, though, it’s perfectly acceptable. The other reason why pro and semi-pro systems by Shure, Sennheiser etc still stick with UHF is its ability to accommodate dozens of systems for large-scale shows. Sure, the ARIAs or a big Broadway show will have someone looking after wireless spectrum as a full time job, but at least they’re guaranteed of high-quality audio that’ll happily travel 100 metres or more. Line6 says the XD-V maxes out at 12 ‘channels’ in a single system, and quotes a reliable 100-metre maximum transmission distance. The great feature of 2.4GHz is that as you keep adding systems, you won’t need to go searching for (a sometimes increasingly scarce) frequency slot, it’ll just spark up. But this isn’t to say that you’re not crowding an already crowded frequency range even further, and often the ‘quid pro quo’ is decreased range.


As an example, we used half a dozen XD-V systems on the AudioTechnology World stage at Integrate a few months ago, and (in an environment seething with RF) we never had a drop out – effectively we were flying below (well, actually, above) the RF radar. Mind you, we had the transmitters and receivers less than 20 metres apart and we weren’t approaching the 12-channel limit – but the setup didn’t miss a beat in the three days of use. GREATER DYNAMIC RANGE

Being digital, the XD-V systems need A/D converters – in this case, 24-bit converters. That’s a lot of theoretical dynamic range. UHF systems use companding to reduce dynamic range for efficiency’s sake. The idea is to recognise that the dynamic range of a vocal performance realistically fits into a 40 or 50dB window so you can dispense with the rest, making the performance easier to transmit… preserve battery life etc. Line6 says ‘baloney’ to all that. It works in the realm of ones and zeros, so once a performance is digitised there’s no need for companding. This results in greater clarity – that’s the promise anyway. The truth is, in most musical performances, and with some console or outboard compression, 40 or 50dB dynamic range is plenty, but I’m happy to buy into the idea that a lack of companding does contribute to a more open/clear sound – in the same way that you get better results when you record a performance in full 24-bit and dither down to 16-bit later. IN THE RANGE

Overall, we had a very positive experience with the XD-V range during Integrate and in the months since. Not only does the system spec out well, with tremendous vocal clarity, but the design is very well thought through. The headset kit, for example, feels right, everything feels robust, the eighthour battery life is good, and it’s very easy to configure. Speaking of configuration, another advantage of going 2.4GHz is the ability to gang receivers. You can nominate one of the receivers as the Master and then the other will slave to the first via co-ax I/O – only one set of antennae required. I have a few minor niggles. The handling noise is more noticeable than with other competing systems and the handheld mic is a tad more cumbersome (the supplied mic clip sagged a bit and needed tightening at times). With a maximum of 12 ‘channels’ per system the XD-V isn’t going head to head with the Big Boys to try and take over the global touring market. But it’ll perform admirably in any club, pub, church or PA rental system. Well worth an audition. I think, like me, you’ll be pleasantly surprised.


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*On selected studio condenser mics

Analogue wireless systems use dual diversity – two antennas and two receiver circuits – to pick the strongest signal. Line6 does something similar in the digital realm – Spatial Diversity. The digital data from both receivers is compared and the one with the fewest errors detected is used. Since the data is received and buffered on both receivers, the decision of which data to use can happen continuously and without any interruption of audio from the switchover. XD-V also employs something called Frequency Diversity, which means it uses multiple RF carrier frequencies to carry different parts of the audio data. Much like antenna diversity, this means any interference is unlikely to be on all the carrier frequencies and can be more easily eliminated as a result.

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KV2 AUDIO SAC2 CONTROLLER Crossover, EQ, dynamics: give your PA some analogue lovin’. Text: Mark Woods

KV2 Audio is a small pro audio company based in the Czech Republic. Founder George Krampera has a long history of designing and building audio equipment including time designing for RCF. In 2001 he and a couple of colleagues from RCF joined forces and KV2 Audio was born. The company aims for the highest possible audio quality in its products, whether analogue, digital or hybrid, and they’re developed around George’s lifelong quest to “reduce distortion and retain the dynamic range and exact detail of the source sound”. That’s a beautiful philosophy if ever I heard one, and striving for this purity of sound has involved technical developments that have made KV2 both an analogue champion and a digital warrior. Primarily speaker manufacturers, KV2 has begun branching out into speaker controllers, and I can see the logic in that. The SAC2 speaker controller is its latest product and although it’s been designed for use with KV2’s ESD speakers it can be used with any speaker system. CROSSING OVER

The SAC2 is a cleverly designed processor that combines a two-way stereo crossover with EQ and dynamics to provide complete control of audio speaker systems. Most current system controllers are digital, whereas the SAC2 is a highquality, latency-free, analogue design. It has outstanding technical specifications courtesy of KV2’s Super Analog technology that places great importance on the accuracy and speed of the underlying electronics. The result is a frequency response of –1dB from 3Hz to 40kHz, 115dB dynamic range and THD of 0.005%. It’s a 1U 19-inch rackmountable unit that’s pleasing to look at and a good example of how an analogue device can be simpler to use with all its functions on display and accessible via knobs or switches. A digital controller could have more functions but LCD displays and nudge buttons don’t compare to real controls for looks or ease of operation. It’s a solid unit and at 3.2kg it feels like there’s a lot inside. The knobs, switches and meters are all good quality and well labelled. It looks simple and intuitive but the control options and operating modes are surprisingly AT 78

comprehensive. Starting from the left there’s a section that contains a gain knob, limiter level knob and a 12dB/octave HPF that switches between 20Hz and 40Hz. The limiter has an active light beside the knob and on the rear panel there’s a switch that toggles both the limiter’s attack and release between fast and slow. The next section contains a four-band equaliser, with defeat switch, that cuts/boosts the signal at centre-points of 40Hz, 450Hz, 2.5kHz and 18kHz. Next to the EQ section there are dual notch filters for the left and right channels. These narrow-band filters are switchable between 6 and 12dB/octave and are intended to either tame PA/room resonances or reduce feedback hotspots. Combined with the four-band EQ section the SAC2 would provide sufficient EQ control for many systems, potentially eliminating the need for the traditional system graphic EQ. DIRECT APPROACH

The output section to the right of the front panel has 4 x five-segment LED meters indicating the Hi/Lo output levels of the left and right channels. Knobs adjust the low and high output levels of both channels simultaneously. The crossover frequency can be switched between 70Hz and 120Hz and there are two switches that defeat the crossover filters and allow full-range signal to pass to both the low and high outputs. Although the SAC2 is designed mainly as a two-way crossover it could be used as a dual full-range system controller by switching both bands to Direct. For instance, the low output could be set to Direct and sent to FOH speakers while the high output could be set to Direct (or via the 70Hz/120Hz HPF) and sent to foldback or in-fill speakers. This setup would work well for bands that mix themselves from either on-stage or side-stage. The EQ and limiting would be across both systems but it’s an interesting idea and adds flexibility to its application options. The rear panel has pairs of high and low XLR outputs each with a phase reverse switch. A Sub mode switches the low outputs between mono and stereo. Each output is driven by KV2’s VHD line driver technology that it claims is capable of driving long cable runs (over 100m) without signal

degradation. The only other control is a switch to engage a low-frequency enhancement circuit that provides boost around 60Hz. This also works in full-range mode and adds another EQ option. OUTDOORS, IN CONTROL

In use, the SAC2 is simple to connect and operate. I used it at an outdoor show with my trusty horn-loaded double three-way and noticed several improvements over my existing crossover setup but the main impression was that it just sounded better… clearer, punchier, very nice. The other improvements were the extra controls at hand. The sub EQ switch added some good solid low end and its set deep enough to get under the often overly-resonant 100Hz –160Hz region. The EQ section had enough controls to be able use it instead of my normal system EQ. Of course, because of my cabling the SAC2 was sitting near the stage with the amp racks and that made it impossible to adjust from the front-of -house mixing position, but once I was happy with the broad settings I found the channel EQ took care of individual tweaks to instruments or vocals. The fourband EQ has been designed to operate without interference between bands and with excellent phase response. It sounds super-clean with quite wide cut/boost shapes well-suited to overall sound sculpting while the additional notch filters were great for controlling the often unruly low/low-mids in my PA. The limiter is designed to be as much a leveller as it is a system-protection device. Its attack and release times

are quite slow and the control knob dials up the desired output level, rather than a threshold. It bites hard if it’s hit with a big level, as it should, but with music playing it’s commendably transparent even with a lot of levelling going on. Back home, I set the system up and did my best to do an A/B comparison to make sure I wasn’t fooling myself about the improved sound. I couldn’t switch from my usual crossover to the SAC2 instantly, or do a blind test, but the SAC2 always sounded better, which is great for KV2 but leaves me with the slightly uncomfortable feeling that my PA won’t sound as good next time I use it. WITH KNOBS ON

These are hard units to fault but I had small concern with how hot it gets underneath the power supply end, almost too hot to touch. KV2’s technical people assure me that even if it does get too hot to touch it’s still within its normal operating range and it’s not going to overheat when mounted in a full rack of gear at an outdoor gig in the Australian summer sun.

And that’s it: simple but powerful, easy to understand and a pleasure to use. The KV2 SAC2 is not a cheap bit of gear but its top quality and very well made. I applaud the fact its analogue. Possible applications for the SAC2 include installations, clubs, churches, theatres as well as PA operators or bands that run their own gear. It would also be ideal for studio monitoring control. Great kit, well done KV2 Audio.

NEED TO KNOW Price $2495 Contact 02 4388 4152 Features Analogue circuitry Super Analog technology Two-way stereo crossover Master EQ plus notch filters Limiter Pros Sound quality Comprehensive controls Ease of use Flexible operations Cons Pricey Gets hot Summary A solid, and great-sounding box of analogue circuitry and features. Your PA will love you for it.

EXPLORE THE AT ARCHIVE – FREE! AudioTechnology online is now packed with thousands of archive articles: reviews, feature stories, tutorials and columns from umpteen years of AT. Best of all, these articles are available free of charge! AT 79


FOCUSRITE SCARLETT 2i2 Focusrite preamps in a giant ‘dongle’. Text: Graeme Hague

NEED TO KNOW Price Less than $200 (estimated street price) Contact Electric Factory (03) 9474 1000 Pros Compact & tough preamps are good quality Cons No MIDI No digital connections. Summary It’s a ‘chuck it in your bag’ audio interface that can deliver great results.

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With half the world intent on making its various flavours of phones and tablets the single device to achieve anything, it’s nice to see companies like Focusrite still releasing small, portable and dedicated gear to get the job done best. You can leave your iPhone off the hook. The Focusrite 2i2 is a USB 2.0 audio interface designed to keep things simple, compact and with good quality. You’re not getting a lot here – just the two Focusrite preamps (which are never to be sneezed at) accessible via a pair of Neutrik Combo connectors with two quarter-inch TRS output channels on the rear. There’s no MIDI or digital connections and the 2i2 is USB bus-powered, so no external power supply.



Driver installation for the 2i2 is very straightforward and you don’t get any virtual mixer or software running in the background. I reckon the 2i2 is a good alternative to a high-quality DI box for playback from a laptop or portable setup. The two preamps and the Monitor knob on the front, acting as a volume control, become almost a bonus. The headphone output is the full-blown quarter-inch jack, which always makes me happy and there is one of those Kensington lock things that I’ve never, ever seen anyone use, but is still a good idea given the Scarlett 2i2 is compact enough to slip inside someone’s handbag.

The main idea behind the 2i2 is a tough, go-anywhere recording and playback interface, and you certainly get that. The anodised, red aluminium case should safely let you hurl the unit across the room and the control knobs are protected – within reason, to keep them accessible – by a small lip on the front of the case. Each preamp has a gain control with a circular LED around it. When the 2i2 detects an input, the LED flashes green. Come close to clipping and it changes to a warning orange – then, of course, “Red!” when clipping occurs. I could watch it all day. An instrument/line switch is supplied for each input. Phantom power is fed to both channels if you select it and there is a Direct Monitor switch to avoid any latency caused by your DAW setup, should that be a problem. I had to run everything really hard – prompting the accusing red LED – including the headphone volume at maximum to get any workable level in Direct Monitor mode, so it looks like you’ll always be better off operating through your DAW to boost levels during recording. The quality of the preamps is, as you’d expect from Focusrite, very clean with a nice yet subtle accent on the top end that adds intelligibility to vocal recordings. Impressive.

What’s also impressive is the bag of extra goodies you get with the 2i2. I’m a bloke’s bloke with a drawer full of blue singlets and I own two – yes, two – chainsaws, so I’m loath to use the word ‘gorgeous’. But that’s exactly what sprung to mind when I saw the Scarlett Plug In Suite with its four plug-ins in their distinctive, shiny red designs. Packaged in the Scarlett Plug In Suite is a Gate, Compressor, EQ and Reverb. The virtual VU meters are great (even the Reverb gets one), although I’m not convinced they’re entirely accurate (who cares, they look purty) – and the controls are clear and precise. You’ll want to use these plug-ins simply because they look cool. Fortunately, they work well anyway.

Actually, it’s small enough to slip inside any laptop bag, too. You can hear the Focusrite quality, it’s USB 2.0 (USB 3.0 compatibility is apparently planned in firmware updates) which is a plus given how hard it is to get Firewire in a laptop nowadays, and it doesn’t need protecting – as long as a few scratches on the case won’t break your heart. The Scarlett is definitely worth considering, if you’re after a portable, rugged interface.

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- MAyer Hawthorne

artist, producer, engineer AT 81



Pearl has been hand-making unique microphones for 70 years, but very few people know about them. Greg Simmons dives deep… Text: Greg Simmons

I’ll bet that most AT readers have never heard of Pearl microphones, and therefore might assume they’re yet another faux manufacturer hiding cheap Chinese crap behind a glossy Western façade. Or, equally as valid, assume that the well-known percussion manufacturer Pearl has cashed in on the ‘own your own’ microphone scam. Both assumptions are quite wrong. Since 1941 the small team at Pearl Microphone Laboratory in Astorp, Sweden, has been quietly hand-making condenser microphones that are so unique almost nobody knows about them. I first heard about Pearl microphones in the early ’90s when a musician passed a handful of CDs across my desk. “Reckon you can make a recording that sounds like these? The engineer uses some strange microphones from Sweden, something to do with rectangular diagrams.” “You mean diaphragms?” “Probably, yeah…” The CDs contained some of the most natural, spacious and dimensional direct-tostereo recordings I’d ever heard; similar to the sound quality provided by ribbons but without the associated dullness and noise. I’ve been curious about those rectangular diaphragms ever since… NOT ALL PEARLS ARE ROUND

Before looking at the review models, it’s worth looking into that rectangular diaphragm. The vast majority of condenser microphones use a circular diaphragm that is AT 82

held under tension, just like a drum skin. The diaphragm’s mass and diameter combine with the tension to produce a single fundamental resonant frequency that exists across all planes of the diaphragm, creating a strong resonant peak in the frequency response that is akin to the note created by a drum skin. The microphone designer has a few options for dealing with the resonant peak. The diaphragm’s diameter, mass and tension can be tweaked to shift the resonant frequency above 20kHz, where it assumedly won’t be audible; this produces a very small diaphragm that has many excellent properties but suffers from higher self-noise. Alternatively, the diaphragm can be damped to tame the resonant peak, but at the expense of sensitivity and high frequency performance – just like putting gaffer tape on a drum skin. Finally, a combination of tweaking the diaphragm’s parameters and applying a nominal amount of damping can be used to turn the resonant peak into something useful; for example, an upper midrange boost to enhance the intelligibility of vocals, or a high frequency boost to compensate for the loss of high frequencies in the air when miking from a distance. Dozens of excellent condenser microphones have been made using circular diaphragms, but that strong fundamental resonance remains an unavoidable fact of physics that often manifests as a shrillness or harshness when boosting the high frequencies more than a few dB.

Pearl sidesteps this problem by using a rectangular diaphragm, replacing the circular diaphragm’s single strong resonant frequency with two considerably weaker resonances: one for the length, and one for the width. Returning to the drum skin analogy given earlier, nobody makes a rectangular drum skin because it isn’t resonant enough to produce a useful musical note; that’s bad for a drum skin, but good for a microphone diaphragm. By tweaking the diaphragm’s length/width ratio and applying a modicum of damping, the result should be a sufficiently sensitive microphone with a smooth frequency response and natural tonality, free of the effects of a single strong resonant frequency. An interesting side effect (sic) of the rectangular diaphragm is that it effectively gives the microphone two polar responses: one in the vertical plane and one in the horizontal plane. When placed vertically (i.e. the usual upright placement) the diaphragm is more sensitive to sounds from the sides and less sensitive to sounds from above and below, particularly at higher frequencies. Such selectivity can be a helpful tool for rejecting unwanted sounds. Placing the diaphragm vertically rejects reflections from the ceiling and floor, a useful feature for making direct-to-stereo recordings where strong vertical reflections confuse the stereo image. Placing the diaphragm horizontally minimises spill from instruments to the left and right, a useful feature when close-miking an instrument such as a double bass with other instruments playing in the same space. The diaphragm rejects spill from the sides while providing its full polar response across the length of the double bass. THE REVIEW MODELS

Two Pearl microphones were provided for this review: the CC22 and the ELM-C. Both are housed in handsome wooden boxes reminiscent of those supplied by Schoeps, with radiused edges and an engraved plaque on the side showing the model number of the microphone contained within. A foam insert separates and cushions the contents, including the microphone (protected from dust in a soft fabric sleeve) and Pearl’s #1930 shockmount, which uses Rycote’s Lyre suspension system. Each microphone is individually hand-tested with MLSSA, and the signed specification sheet shows its serial number, frequency response, sensitivity and self-noise.

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The CC22 is a cardioid condenser microphone using a rectangular dual diaphragm measuring 30mm x 20mm. The supplied specification sheet shows a sensitivity of 18mV/Pa and a self-noise of 12dBA; both respectable figures. The measured frequency response remains within 1dB of flat from below 40Hz up to 3kHz, exhibits rises of up to 2dB between 3kHz and 9kHz, remains within 1dB from 9kHz to 17kHz, then rolls off to -5dB at 25kHz. Very promising. The ELM-C is a cardioid condenser microphone using a considerably longer rectangular dual diaphragm measuring 70mm x 11mm. The specification sheet for the review unit shows an impressive sensitivity of 28mV/Pa and an equally impressive self-noise of 9dBA. Its measured frequency response begins 2dB down below 40Hz, is essentially ruler-flat from 60Hz to 7kHz, rises to +2dB at 10kHz, deviates within ±2dB up to 20kHz, then rolls off to -7dB at 25kHz. Also very promising, and possibly well suited to distant work.

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The CC22 and ELM-C offer very agreeable specifications and tick all the right boxes in the look and feel categories, but how do they sound? To find out, I tested both microphones during a number of sessions at the Australian Institute of Music. In most of the tests I level matched the Pearl microphones

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No information is given regarding the distance from the source that the frequency response measurements were taken from, but the extended low frequency response suggests about 30cm. Both microphones are satisfyingly heavy for their size, and include solid XLR inserts with gold-plated pins. The Pearl logo and model number are etched into the front with gold lettering, the serial number is etched into the back. A red LED mounted immediately beneath the diaphragm on the back of each microphone illuminates when phantom power is supplied – clearly visible to the engineer but sufficiently inconspicuous to the artist. The build quality is good and solid, the design is sensible and acoustically unobtrusive, and the high gloss black lacquer finish is so lustrous out of the box that merely touching it feels like a sin.

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against established industry standards to provide points of reference that many readers would be familiar with.

the ELM-C but with less low frequency energy, and without the U87’s excessive roominess.


As a final test, I pitted the CC22 and ELM-C against each other and a U87 on a close-miked male vocalist performing a soul song a capella. Both Pearl microphones made the U87 sound gritty, veiled and dated. The CC22 delivered a very appealing result but the ELM-C stole the show with a sound that was remarkably ‘present’ and full-bodied with nowhere near the amount of proximity effect I had anticipated after the earlier grand piano test.

This is a very natural sounding microphone that is highly suited to close-miking acoustic instruments. Placed perpendicular to the soundboard of a violin at a distance of just over a metre and compared to a Neumann KM184 in the same position, the CC22 sounded subtly sweeter and rounder, less ‘steely’ on edgier high notes without sounding dull, and offered a touch of the instrument’s body and resonance that was sorely missing in the KM184. It tracked the aggressive transients of Bartok-styled pizzicatos with ease, and held its composure through harmonically complex passages that left the KM184 sounding thin and etched. The CC22 produced an excellent sound that did justice to the instrument, clearly superior to the KM184 yet remaining tonally compatible with it.

NEED TO KNOW Price CC22: $1683 ELM-C: $2310 Contact Professional Audio Technology (02) 9476 1272 Pros Natural uncoloured sound. Useful polar response variations in horizontal and vertical planes. Well made and presented. Good specifications. Appropriately priced for what they offer. Cons None, provided the user is aware of the differing polar responses in the horizontal and vertical planes. Summary Pearl’s CC22 and ELM-C rectangular dual diaphragm microphones offer a uniquely natural and uncoloured sound, similar to that provided by a ribbon microphone but without the associated dullness and noise. They deserve a place in every microphone collection.

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In contrast to the violin recordings, I tried the CC22 on two different electric guitar recordings. The first was a heavily distorted sound created by a Fender Squire through a Fender 15 amplifier. Placed 30cm in front of the cabinet the CC22 gave a smooth and clear rendition with no apparent glare or harshness, bringing the microphone’s subtle frequency response rises between 3kHz to 9kHz to the fore. (One would have to be careful when applying EQ to avoid pushing the sound towards thinness.) In comparison, Neumann’s U87 in the same position sounded as if it was going through a band-pass filter, with no low or high frequency extension. It was duller and very nasal, but offered more chugga-chug grunt. The second was a relatively clean-sounding Fender Telecaster through a Roland JC120. Placed a few centimetres in front of the cabinet and slightly off-axis to one speaker, the CC22 sounded smooth and uncoloured, prompting another engineer on the session to remark that it reminded him of Royer’s R121 but with better high frequency extension. THE ELM-C IN USE

This is a very natural sounding microphone that excels at distant work; it is really in its element at distances of 30cm and beyond. It also works well up close but, due to the length of the diaphragm, its relative angle to the sound source becomes quite critical. The first test was on classical guitar, with the ELM-C placed about 60cm from the point where the neck joins the body and focused where the strings were plucked. Tweaking the microphone placement and angle achieved a detailed, warm and even sound that, like the CC22 on violin, was never harsh or thin and was clearly preferable to a KM184 placed in the same position (for what such a comparison is worth). For the second test the ELM-C was placed just outside the crook of a small grand piano and aimed towards the centre of the strings. The resultant sound offered full and round low notes and smooth but precisely detailed high notes, even on the harshest short-stringed high notes. A very good result for a tweak-free initial placement – instant piano sound! It was, however, in danger of sounding bloated; I suspect that moving any closer would’ve yielded too much low frequency energy due to the proximity effect. Interestingly, a Neumann U87 placed in the same position sounded unacceptably roomy and distant, and considerably lacking in low frequency energy relative to the ELM-C. Placing the CC22 in the same position produced an acceptable sound with the same high frequency character of


Throughout this review I have tried to resist comparisons with ribbon microphones based purely on their visual similarities. It’s an easy trap to fall into; at first sight most engineers and musicians assumed the review units were ribbons. Although they are very different technologies, they do share at least one common trait: a very natural sound due to a lack of any strong resonances in the midrange and high frequency spectrum. Importantly, however, the Pearl microphones do not suffer the perceived dullness and high noise floor associated with ribbons. In that respect, the rectangular diaphragm nicely straddles the gap between ribbons and circular diaphragm condensers without making any concessions to either technology. For this reason alone, every microphone collection should have at least one rectangular diaphragm condenser microphone. The CC22 and ELM-C are both excellent cardioid microphones for situations that require a natural and uncoloured sound without sacrificing detail and high frequency extension, and without risking a high noise floor. The CC22 is applicable for all close-miking duties that would normally be undertaken by a condenser microphone, and delivers a sound that subjectively lies somewhere between a Royer R121 and Neumann’s KM184. The ELM-C is a very interesting microphone that shares many of the CC22’s characteristics but is best used at distances greater than 30cm or so, producing a full-bodied and present sound where other directional microphones would start sounding thin and detached. Astute readers may have noticed a recurring observation throughout the tests described above: whenever one of the review microphones was compared with a traditional condenser, the latter always had more room sound. I attribute this to the rectangular diaphragm’s tendency to reject sounds arriving from above and below, as discussed earlier. Without the additional vertical reflections, the rectangular diaphragm always sounded closer and less cluttered than a circular diaphragm at the same distance. With that in mind, I’m very keen to try a matched pair of rectangular diaphragms on a string section, choir, chamber music ensemble or one of my field recordings in a Himalayan monastery. The rectangular diaphragm’s uncoloured tonality combined with its rejection of interfering vertical reflections is bound to yield a direct-tostereo recording that is natural, spacious and dimensional – just like those CDs that triggered my initial interest in rectangular diaphragms all those years ago. I look forward to adding a twin set of Pearls to my collection in the near future.

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Studio One instantly won friends for being lean and mean. Can version 2 meet user demand and stay bloat-free? Text: Graeme Hague

Bear with me during a long-winded and waffling, yet relevant and (no doubt) riveting introduction here. It’s not easy endeavouring to take on the big players by releasing a new flavour of DAW and we need some historical perspective here. As we reported last year following the original release of its new Studio One software, Presonus made a significant impact on the market and quickly attracted a faithful user base. Studio One was touted as being ‘bloat free’, meaning Presonus had been very particular about the features included in Studio One and it tried to avoid the more esoteric functions the competitors’ software offered which, frankly, only confused the hell out of most of us anyway. A simple plan’s always a good one – or, in this case, a simple application is always a good one. However, this policy must have caused a few headaches for Presonus when it decided to release Studio One Version 2... and yes, the marketing department must have chewed a lot of fingernails just agonising over what to call it. In the process of creating a better, more comprehensive DAW has Presonus been forced to chuck the ‘bloat-free’ policy in the bin? Making things even more difficult, Presonus enjoys a good relationship with its users via a relaxed, flame-free forum with Presonus administrators always active. As you can imagine the obligatory “What do you want to see in the next version of Studio One?” thread is chockers with opinions from users who are convinced that their requirements are the best, most brilliant ideas everever and it’s a no-brainer that Presonus should completely rewrite the entire program to indulge them – functions they use every day, and it’s incomprehensible that others AT 86

actually don’t. Odd folk, in other words. So does Presonus honour that good relationship and listen to the roar of a zillion feature requests or not? Does it mean the bloat-free DAW has gotten a little tubby around the waistline with Version 2? STABLE INFLUENCE

As you’d expect, Studio One offers an impressive audio workspace with good plug-ins and comprehensive editing functions. Previously it lacked stuff like pitch correction and beat detection (can you see where this review is going already?) and the basic Artist version didn’t support thirdparty plugs, while the Pro version did. MIDI functions were on a par with... hmm, let’s say, ProTools, meaning any aspiring MIDI programmer would be happy unless they need the MIDI editing extremes provided by the likes of Cubase, Logic or Sonar. The included VST instruments are pretty cool and certainly usable. The pianos are nice pianos, the organs are vintage and the synths are trippy – but let’s be honest, they ain’t Absynth or BFD. One thing that got a huge thumbs-up was S1’s stability. With S1 in full flight, but minimised, you could still run emails, monitor Facebook and download complete series of Family Guy and Studio One would cheerfully chug away unfazed in the background. It was awesome. For DAW users who were tired of their software crashing whenever things got only a bit hectic, Studio One was a breath of rock-solid fresh air. Now you’ve got three versions, Studio One Artist, Producer and Professional. Everything new comes in Professional, so for the moment just assume that’s what we’re talking about and I’ll explain the differences in full later.


You should have guessed by now that S1 V2 does come with pitch correction and beat detection. Presonus has chosen to join forces with Celemony and its Melodyne software, but instead of just tossing in a freebie plug-in with any purchase of S1 V2, Melodyne Essential has been completely integrated with Studio One. Melodyne is initialised through a right-click menu function to ‘Edit with Melodyne’ and other processes such as rendering any pitch-corrected audio are also done inside S1 dialogues. Presonus is claiming this close integration is a first (although others will undoubtedly follow). You might think that it doesn’t make a great deal of difference except that processes like Undo and Bypass, for example, are part of the S1 menu hierarchy. Then again, pitch correction plugs can be a little daunting the way they take over audio events and with the S1/Melodyne setup it certainly feels like you’ve got more control over what’s happening. Events can be smoothly moved, copied and pasted, or edited in a variety of ways and any pitchcorrection processing happily goes along for the ride. I’d suggest too that Melodyne was a good choice by Presonus. Melodyne’s simple (there’s that word again) approach to manipulating audio with those unique ‘blobs’ of sound take a lot of the rocket science out of pitch correction. TRANSIENT DETECTION

The Transient Detection feature in S1 is quite understated in Presonus’s ‘what’s new’ blurb. It is, in fact, a whole rack of editing functions under the Audio Bend menu that allows pushing, pulling and generally bullying errant audio performances into some kind of proper timing–otherwise called warping, stretching... whatever, take your pick. The audio is moved between markers with that concertina-like action to create seamless shifts onto the correct beat. Studio One calls it Bending and does it well. Transients can be automatically detected, the detection level can be adjusted or you can add and delete your own markers. This in turn allows you to apply quantisation and even groove effects to your audio. A neat trick is that any audio stretched between two markers turns alarmingly more red in colour as you increase the distance – an indication of your growing desperation, perhaps? – while compressing events are shown as comforting, soothing green. Studio One Version 2 has introduced comping with layers for each track. Again, it isn’t anything radically new, although Presonus’ system is easier than most: you can select a range of any ‘take’ and double-click to add it to the primary track. Crossfades are dealt with automatically as is a colourcoding feature that helps you instantly refer back to which layer that particular section was sourced from. For the record, creating layers and comping was probably the loudest, most insistent feature request on the forums and it’s great to see Presonus respond accordingly. TWEAKS & FIXES

Likewise a lot of minor tweaks, too numerous to list here, have been applied across the board to various Tool behaviours, plug-in windows and such. Plainly, Presonus has listened to the feedback and picked the wheat from the chaff – or obviously made development decisions for itself. Most users should be pleased with what’s been improved and what hasn’t been changed. One niggle for me is how the mixer section, which normally sits at the bottom of the window, can be detached and stretched to full screen, yet the size of the actual faders, which I’ve always found a bit short, doesn’t increase (even though the input fader does). If you don’t have many plug-ins or sends running, the result can be a lot of empty, grey panels in the mixer that offer nothing. There is a new Narrow mode where the panels are replaced by long meters, which is interesting to watch, but I still don’t get my longer faders. GUI POLISH

There’s one enhancement that has divided the larger forum community, creating a thread rampant with emoticons like a bad, smiley rash. The GUI has been given a wash and polish. Fonts have been sharpened, bold sub-headings have been added, function buttons have either disappeared or appeared, mouse highlighting has become... erm, higher. Studio One doesn’t have any scheme or ‘skin’ options, so users who upgrade to Version 2 can’t avoid these changes. Some people aren’t happy. Myself, I’m well pleased with the improvements. One of my strongest criticisms of S1 was that the whole AT 87

A look at the v2 Mix Window – with its more engaging GUI. Ampire XT (top right) packs 13 different amp models and a choice of 10 cabinets, while Open Air (bottom right) provides convolution reverb – the separate IR Maker plug-in allows you to load your own impulse responses and create custom spaces..

NEED TO KNOW Price Studio One Artist 2.0: $111 Producer 2.0: $222 Professional 2.0: $440 Upgrade pricing available. Contact National Audio Systems: 1800 441400 Pros Very stable Still no ‘bloat’ in the features Professional version has the excellent Project page and integrated Melodyne Essential Cons Onboard VST instruments are okay, but for par excellence invest in something like Kontakt 5. Ampire XT is an exception – it rocks! Summary Presonus has kept the faith, improving and enhancing Studio One without adding any unnecessary features. It’s a solid workhorse that focuses on the tools you need.

AT 88

look of the software, particularly the mixer, was verging on the bland and while I agree that a DAW isn’t supposed to resemble Level 3 of a Sonic The Hedgehog platformer, I’d argue that a lively workplace is inspiring. After all, a real control room with a glittering, flashing SSL console is exciting, right? Not that Presonus has added any pointless eye-candy, but rather it’s managed to make the GUI more engaging... it’s difficult to pin-point, but S1 has definitely lost its blue-grey Vaseline-d appearance and now looks professional and more inviting to work in. AMP’ED

A few of the new effects in v2 deserve a mention. Studio One’s own guitar amplifier emulator, called Ampire, was rather underwhelming in version one. Hey, it worked, and the limited sounds were okay, but it was never my first choice for converting my fumbling fretwork into Jimi Hendrix-like genius (I need all the help I can get). Now it’s Ampire XT with 13 different amp models and a choice of 10 cabinets, all of which can be assigned to either the A or B channel, then you’ve got eight stomp FX essential to shaping a guitar ‘sound’. That’s the kind of help I need! Also new in the effects rack, the Open Air convolution reverb should satisfy the reverb aficionados and if it doesn’t, there is the IR Maker plug-in which allows users to load their own impulse responses and create custom spaces. PROJECT POLISH

Back to the different versions of S1 V2: Artist is the basic software that still won’t allow third-party plugs and doesn’t have Melodyne. Producer does support AU/VST plugs and MP3 import/export, but there’s no pitch correction. Professional gives you the lot and includes the innovative Project page – which is well worth revisiting here.

The Project page is a mastering environment, a separate window altogether that will run in the background of any Song page. Here at AT we’re always stressing that professional mastering isn’t something you can achieve with a plug-in, it requires specific skills, equipment, and experience, so we encourage taking your tracks to a proper mastering facility. But there are many occasions when this isn’t practical and that’s where S1 can save the day. The Project page allows you to drop your mixed down song files onto a proposed timeline of a CD. You can apply any or all the effects you like, such as compression, EQ – the usual suspects – to each individual song to bring them up to par with each other and you can dial up similar plug-ins to a master output too. You have complete control over the final CD or, if CDs aren’t your thing, a method of ensuring continuity of sound across a body of tracks (ie. an EP, album, soundtrack, you name it). If you create a complex project filled with tracks, plugs and a unique timeline arrangement, then suddenly realise you left the kick drum out of one song (you idiot!), you can open the Song page and make amends before telling S1 to ‘update’ the appropriate Project. The underlying, source mixdown file is replaced without disturbing all the other Project settings. Clever. NO BLOAT

Studio One Version 2 is a tight, professional DAW that isn’t overweight with obscure, extra menus or functions only a tiny minority need. And what it does have works well on both PC and Mac. It’s still rock solid, too. But don’t muck around – get the Professional version. The Project page is worth the extra dollars alone.




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All the T-Pain-Andrews-Sisters power of VoiceLive in your FOH rack. Text: Guy Harrison

NEED TO KNOW Price $1399 (inc MP75 mic) Contact Amber Technology 1800 251 367 Pros One-box vocal solution. Great user interface. Massive preset library. Cons Won’t suit all types of music. Wall wart power supply. Summary The TC Helicon VoiceLive Rack is a compact all-in-one vocal solution. With a dazzling depth of programming it certainly rewards the tweaker, while the massive preset library and search wizard will do the trick for everyone else.

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TC Helicon a division of TC Electronic dedicated to ‘your voice’ introduces its latest offering, the VoiceLive Rack. Bringing all their technologies into one box is not a new concept for TC Helicon as this was done previously with its VoiceLive Extreme. While that product was performance orientated with its floor mounting and footswitches, á la a guitar stompbox for the vocalist – this time TC Helicon brings its vocal technologies to the FOH mix position with the VoiceLive Rack.

Kanye preset it’s inserted after the lead vocal on the send to the other effects, ie. Delay doing some ¼ & 1/8th-note triplet delays. The result? Hey presto, I am Kanye! [You only need to check out Guy’s DIY preamp video on the AT YouTube channel to observe what a miraculous transformation that is – Ed.] Complete with big metallic sounding delays that duck while I sing, then pop up at the end of my phrases. And I can’t sing out of tune no matter how hard I try. Okay, maybe if I really try…

Designed to be used live as a fully featured vocal channel, the VoiceLive Rack packs a lot of punch into its 2U chassis. Although the VoiceLive Rack can be configured as a ‘send and return’ effect, its true power lies when utilised as a vocal channel using the onboard mic pre. The reason being that where the VoiceLive Rack excels is in its pitch-based effects which, quite simply, cannot be used in a send and return configuration. While on the subject of pitch, the VoiceLive Rack also boasts a ‘guitar/music’ input from which the harmony generator can take its pitch cues. Very nifty indeed! Add to this Touch Control and a supplied MP-75 microphone with an onboard button for controlling the effects and we have a very unique offering.

After 10 minutes of wading through many harmony-style presets and finding myself singing ‘When I went down to the river to pray’ from the Coen Bros O Brother, Where Art Thou? film, far too many times, I came to the conclusion that the presets are very well programmed. The harmonies are often convincing, if slightly overcooked – I guess TC Helicon doesn’t want to leave anything in the locker that may impress.


Right. Time to switch this baby on, plug the MP-75 microphone in, and cruise the presets. Instantly, TC Helicon capably shows the wares of the VoiceLive Rack. The very first preset is called ‘Another Brick’ and, well, who could resist belting out a few bars of this seminal classic! This preset uses the Double, Harmony and Reverb blocks and while slightly overdone was convincing. The next preset: Kanye Lockdown. Here we get some trademark Kanye action utilising three processing blocks. HardTune is taking care of the very deliberate Cher-style pitch correction, with some Transducer to get the highpass 456Hz filter processing. This processing is able to be inserted in different routing positions. In this particular

I found one very handy feature to quickly modify a preset: tweaking any of the small rotary encoders brings up a screen that allows you to balance the ‘Voices’ (ie. harmonies) and the Reverb/Delay levels against the dry vocal. Sadly, the latter are only available on the one knob – separate reverb and delay levels would be very useful. That said, this does provides a quick way to knock the effect levels back a little on the fly. These controls are global, so flicking to the next preset keeps the settings – great if using the VoiceLive Rack live. HARMONY TRACKING

Next I had a quick play with the harmony tracking feature. The VoiceLive Rack has the ability to track an input, be it guitar, aux or MIDI and use this as the reference for its harmonies and pitch correction. Sound fraught, but in practice the chord recognition was very fast and the ability for the harmony engine to track from something other that your voice opens up another world of possibilities. The Guitar Input can be routed to the output and has its own

UMOD: Flange, phase, chorus etc HARDTUNE: The effect you’ll know as the Cher/T-Pain type. TRANSDUCER: Filter and Distortion based effects like ‘megaphone’.

REVERB: From Plates to Halls and everything in between HARMONY: Vocal Harmonies up to 4 Voices.. MIC CONTROL: Used to assign a function to the button on the supplied MP-75 microphone.

DELAY: Delay effects from slaps to ping-pongs, etc. DOUBLE: Vocal doubling up to four voices. RHYTHMIC: Chopping, stutter and panning effects.

independent processing by way of reverb, compression, UMod & three-band EQ. The aux input can also be routed to the main outputs – handy for backing tracks etc, in a one man band or duo. DECLARED INTEREST

I’ve got to confess that I’ve been dead keen to get my hands on the VoiceLive Rack for some time as I’ve been doing some FOH work for a band whose album is heavily driven by harmonies and delays. I was looking to the VoiceLive Rack to bring some of those flavours to their live shows. So off into the VoiceLive programming world I went, searching through presets and modifying them to suit my needs. With 238 presets on offer the Wizard button is a welcome relief. Here you can refine your preset search using keywords such as Harmony Below, Octaves, Long Ambience, Tap Delay, HardTune etc. With the use of up to three keywords you can find what you’re chasing quickly. Recalling any preset out of the revised preset list is easy with a second press of the Wizard button. My list of preset candidates selected, it’s time to get under the hood of this beast. Editing is simple. Copy your preset to an empty location, and turn off the effect blocks you don’t want. Touch and hold any block you want to edit and the Edit window appears. Navigate to the relevant parameter line with the big knob and tweak the setting with the small knob. Simple and effective. The depth of programming within each block is extensive – you’ll find all the settings you would expect and a few extras. The harmony block is particularly worthy of mention with settings for Voice Gender, Humanisation, Smoothing (which allows for pitch correction on the harmony voices), Portamento (glide time), seven different vibrato styles,

settings for choir styles and voice doubling, and you start to understand how this unit goes about its work – there’s plenty of depth. IN STEP

Then it’s time to save your tweaked preset in the new location. A handy feature is the ability to have steps within each preset. The idea is that, say, you have a particular song where you want to use a distorted megaphone voice for a chorus and at the same time reduce the reverb level. Rather than have to create a separate preset you just hold the Step button within your current preset and a step is created. Within this step you can now turn on the Transducer block by touching it, open the Reverb Edit page, reduce the reverb level and store again. Now you can step back and forth between the two steps, turning the Transducer on and reducing the reverb level with the touch of a single button. Up to 10 steps are available and you can change as many settings as you like within each step. Very powerful stuff. The VoiceLive Rack’s performance is outstanding. I must admit, I was apprehensive at the thought of touch controls but in practice found them to be excellent. Being able to touch the Delay block to send phrases or words to the delay was seamless. Touching in the Doubling and Harmony for choruses was a cinch. I really can’t fault this unit. Admittedly, this style of programmable effects will not find a home in all styles of music but if pop or hip hop is your bag (and you’re not afraid to get your hands ‘dirty’ with some editing) then some outstanding results can be achieved. Ideally, I’d love the touch controls to be backlit for low-light FOH conditions. Which, I guess, goes to show that you can have everything and still want more.

THE MP-75 MIC The supplied MP-75 mic is a worthy contender for any job. When compared with the venerable SM58 I found it to have a slightly bigger presence lift. The Mic Control button on the mic makes for some cool performance options although it unfortunately can’t be used to step through presets (a footswitch sold separately can). Setting it up as a momentary send to the delay was fun for sending single words off into delay land providing some useful performance control.

AT 91



A go-anywhere surround recorder for under $300? Zoom zoom. Text: Greg Walker

Zoom has been doing the portable recorder thing for a long time now. Through years of development and refinement of the basic concepts, as well as throwing in various bells and whistles, its pretty much figured out how to tick all the boxes in terms of ergonomics, features and sound quality. Zoom’s latest offering, the H2n, is not the smallest recording device ever made but instead offers an impressive array of features including X/Y, mid-side and surround recording as well as handy extras such as a built-in speaker, tuner and metronome while being easy to operate and pretty light on the wallet by today’s standards. POCKET ROCKET

NEED TO KNOW Price $299 Contact Dynamic Music (02) 9939 1299 Pros Great sound quality Surround and MS recording modes open up many possibilities Easy to operate and manage files Steinberg Wavelab LE software is a bonus Cons Four-channel recording modes can’t record above 48kHz Fatter than a cassette! Summary Zoom’s H2n sounds great and is user-friendly but boasts a considerable power and versatility. Surround recording modes are definitely the crowd pleaser here along with relatively robust and professional styling. A fantastic all-round portable recording solution at a modest price.

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At 67.6mm x 113.85mm the H2n is about the length and width of an old-fashioned audio cassette while curving out to a relatively tubby 42.7mm in depth and weighing in at 130g without batteries. It’s compact enough to sit inconspicuously on a desk during an interview while exuding a modestly professional aura thanks to the shiny black plastic finish and all-concealing black steel mesh microphone cage. Personally, I like this look better than the exposed microphone style, where I immediately imagine all the ways I could accidentally damage the device beyond repair. The unit runs on two standard AA batteries for up to 20 hours when recording at 44.1kHz, which is generous. Durability is an oft-overlooked aspect of the mini-recorder market but the H2n seems sturdy enough and during the couple of months I’ve been using it there’s been no reliability issues whatsoever. Having said that, I wouldn’t be game to drop it on a hard floor as there’s a lot of electronics lurking just below the surface of the H2n. CONNECT FOUR

One of the aspects I really like about the Zoom’s latest offering is the front of the device: a screen that keeps you informed of all vital signs and a large Record button with a red dot on it – nice and to the point! The left side is peppered with I/O ports: minijacks for external mic input, headphones and dedicated remote port, mini USB for computer data interfacing and a +/- volume control. On the right side of the unit there’s a menu button, multi-function transport control (play, fast forward, rewind and stop), a generously-proportioned continuously variable record level dial and a power switch that can be latched to prevent accidental power-downs at inappropriate moments. On the top of the unit is the real audio candy: an illuminated switch that selects any one of the four available microphone recording patterns. These are stereo X/Y, stereo MS, fourchannel surround or four-channel internally summed to stereo. The screen itself is a tastefully backlit number which in recording mode offers information on battery status,

microphone pattern, recording time, sample resolution and bit-rate, any limiting or compression settings as well as an easily visible and responsive LED meter and a clock. WHAT’S ON THE MENU

The H2n’s features run quite deep once you head into the menu and start tooling around. You’ve got a lot of options for compression and limiting setups, low-cut filtering and optional automatic recording gain adjustment. You can tinker with the mid-side spread as well as parameters such as adjustable playback speed, raw or pre-mixed MS processing, and the list goes on. File management is very straight forward. Your computer sees the H2n as a standard USB storage device and the files (stored internally on a standard SD or SDHC card) can be downloaded in a jiffy. The H2n will record at up to 96kHz resolution in stereo modes and 48kHz in four-channel modes as well as in compressed MP3 format. As an added bonus the unit ships with Steinberg’s Wavelab LE software. CAPTURE & RELEASE

I’m quite fond of the H2n. Firstly, the sound quality is excellent, and with a little tinkering of the record levels and limiter settings you can capture everything from live gigs to quiet atmospheres with relative ease. Also the continuously variable record level control sets it a peg above the dreaded ‘auto-record level only’ devices. Secondly, the MS and surround recording modes are just fun! I’ve done some ambient recordings walking around rooms and outdoor locations and the representation these modes give you of physical spaces can be genuinely stunning. If you’ve recorded in Raw mode you can then of course drag it all back into your DAW and tweak it to your heart’s content. I’ve recorded interviews, rehearsals, impromptu kitchen jams and birds singing in the backyard, and in each of these cases the surround capabilities of the H2n mean I’m getting a better representation of what actually happened than I would with a stereo-only device. Finally, I find the little fella very easy to use, only delving into the manual and menus now and again when I need to tweak something for a particular scenario. In general, I just grab the H2n and hit the Record button – a great thing in my book and testament to the oft forgotten fact that although many machines like this offer an abundance of features, they should help you concentrate on what you record, not how you record it. And in this respect, Zoom has hit the bullseye with the H2n.

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AT 93


ACTIVE PICKUP LINES Are you an active or passive mix device? Text: Graeme Hague

Passive and active stuff – it’s pretty straightforward. There are passive or active speakers, passive or active pickups in a guitar, passive or active mixers… “Ah, hang on,” I know you’re saying by now. “Don’t you mean analogue or digital mixers?” Nope, I’m definitely talking about passive or active mixers, but the only bit of hardware that’s really involved here is the two-legged homo sapien ‘hardware’ device standing behind the mixing console. That’s you – yes, you. Here’s an extreme example: The scene was an infamous Sunday session at a North Queensland pub. This was an upstairs bar that was open for five hours or so on Sunday arvo’s and they always had a great circuit band playing (it was a bastard of a bump-in up the fire escape stairs, by the way), the beer – XXXX unfortunately – was cold, and the punters were willing. This particular Sunday was no different. The band was loud and pumping, the place was heaving and the crowd was going off like a Chinese fireworks factory. Amidst all this chaos the mixing guy behind the desk was reading a paperback novel. I couldn’t believe it. While the lead singer was belting out Roxanne for all he was worth, turning blue in the face, the sound engineer was calmly thumbing his way through a book. Now, I figured there were three possible reasons for this. First, maybe it was a really, really good book and he simply couldn’t bear to put it down. It happens, right? Second, perhaps this guy had a brain the size of a planet and to be mixing a very loud band through a scary-type PA while being surrounded by excitable, semi-clad Nordic backpackers simply wasn’t stimulating enough and he needed to occupy the other side of his noggin with a good book to fend off stupefying boredom. Yes, unlikely, I agree. In fact, the most likely explanation is the third option, thanks to Occam’s

AT 94

Razor – although, why shaving toiletries even get a mention in that theory is always a mystery to me. The answer is that the novel-reading sound engineer was completely pissed off with the band and his role as mixer, and he was doing less than going through the motions. The care factor was below zero, but because this was a touring crew, the band was stuck with this bloke for the moment. So this guy was a seriously passive mixer – uninvolved in the band, the music – and the mix. Get the idea? SWITCHING ON

Okay, it is an extreme example, but recently I had the opportunity to wander around a festival where different PA rigs, different acoustic environments, different acts and different operators all abounded and it was interesting to see how passive or active most engineers can be. I reckon there are three types of live mixing engineer. The first is an engineer who sets up a good mix and apparently that’s when their job ends. There’s no denying the soundcheck sounds just fine. For the rest of the show their role is to baby-sit the desk, guard it from drunken punters and turn on the iPod during the breaks. They don’t take any responsibility for the dynamics, the energy or creativity coming off the stage. The musicians have effects pedals for that kind of thing. Fiddling around with volumes is not the audio engineer’s problem. That’s not just passive, that’s nearly comatose – and lazy. Next we have the engineer who is more like the band’s safety valve. Once the soundcheck is complete and everybody’s happy, the aim of the sound engineer is to somehow maintain that status quo, no matter what. From that point on the mix is all about damage control and if you don’t have to do too much, it’s been a good night. On a bad night you’re like a bucket of cold water being poured on

the band’s enthusiasm, stomping out spot-fires of excess by cranking up compressors and limiters, and slapping down faders. You’re just a living, breathing anti-feedback device. GET INVOLVED

An active sound engineer is listening to every song, getting involved with all that’s happening and working the faders like it’s an instrument, too. They’re closely in sync with the band for every lick, riff and fill, and if they’re not familiar with the music such as in a festival environment, they do their best to anticipate things. You take risks (within reason). You never stop chasing the elusive perfect mix, you never assume the band will take care of dynamics for themselves – you’re always ready for anything. My best friend during a live mix is the Tap Tempo button on the delay effects. I’m constantly updating the tempo of the delay with the timing of the song. The faders for the effects returns are next, riding them up and down with dramatic moments of the vocals. Making solos loud and clear is crucial and solo performances can come in many guises, not just guitar-god moments. During a studio mixdown, solos get heaps of space and clarity, so why not during a live show? Yet so often you see a live audio engineer assume the musician will provide that boost for themselves, probably through a pedal of some kind. For some of you this is surely teaching your grandmother to suck eggs. For others it’s hopefully got you thinking – are you a passive or an active sound mixer? When you’re doing a live gig are you playing it safe or playing with the band? Have you got time to read a good book? If you have, at least make it one of mine – and don’t buy it secondhand, you scrooges. I need the royalties. Graeme Hague is an author, an audio engineer and a DIY house builder – he likes to stay ‘active’.

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AudioTechnology Issue 85  
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AudioTechnology Issue 85