AudioTechnology Issue 82

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THE STEP-BY-STEP GUIDE To Building Your Own JLM Mono Preamp ISSUE 82 AU $7.95(inc gst) NZ $10.95 (inc gst) File Under ‘Music’

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WORKSHOPS PANEL DISCUSSIONS MASTERCLASSES LATEST GEAR ON THE BENCH - LIVE! Everything for the AudioTechnology reader, in action, in the flesh.

REGISTRATION NOW OPEN HOW TO BE PART OF AT WORLD & BOOK SEMINARS 1. Register for Integrate online at 2. All AT World sessions are free – turn up to Integrate, then it’s first in is best dressed. 3. Other sessions, like the masterclasses, AFTRS sessions, Stav, Simmo etc require a ticket. Hit the ‘Seminars 2011’ menu button on the Integrate homepage.


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RETURNS MORE INTEGRATE NEWS Stav Returns!: Mike Stavrou will be back in 2011 demonstrating key ear-opening concepts from his Stav’s Word column and top-selling book, Mixing With Your Mind. Simmo Returns!: Greg Simmons will also be reprising his day of Studio Fundamentals, including: ‘Getting it Right the First Time’, Monitoring Fundamentals’ and ‘Microphones – Choosing & Using’. CHOOSE YOUR WEAPON: DAW MASTERCLASSES Regardless of your weapon of choice, we’ve got you covered with ProTools, Logic Pro and Cubase masterclasses. Power users take note: there’s nothing entry-level about these sessions. ProTools Drum Mastery: Brent Heber (Avid-certified instructor at UTS:ProSchool and regular contributor to AT) will help you get the most out of Beat Detective and Elastic Audio – a must if you work with live drums or loops/samples. Logic Pro Advanced Techniques & Workflow: Paul Najar (Apple-certified master trainer, UTS:ProSchool Instructor, renowned musician, engineer and producer) delves deep into phase accurate time correction of live drum recordings using Flex, mixing electronic and acoustic percussion using multiple outputs in Ultrabeat and EXS24.


Cubase Expression & Automation: Yamaha Music Australia’s Mick Hughes will take you on an intense 90-minute trip through some of the latest and most powerful features of Steinberg’s Cubase 6, as well as into some of the hidden depths of this DAW monolith. AFTRS’ 4 BIG SOUND-FOR-PICTURE SESSIONS Right next door to Integrate is the Australian Film, Television & Radio School, where you can access these four great sound-for-picture sessions. Mixing in Surround: AFTRS Head of Sound, Chris McKeith, explores and demonstrates aspects of making and mixing 5.1 surround soundtracks for film and TV. Advanced Mixing in ProTools: AFTRS Head of Sound, Chris McKeith, examines features and techniques using ProTools and ICON consoles for mixing 5.1 soundtracks. How Film Music Works: AFTRS Composer and Head of Screen Music, Martin Armiger, guides you through how to successfully marry the worlds of film and music. Introduction to Location Sound: Aspiring location sound recordists will learn the key concepts, hear and see real-life examples of where location sound has gone right and where it’s gone very wrong. Presented by Rod Pascoe.

in association with

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Editor Andy Stewart

Stem-sell research.

Publisher Philip Spencer

Text: Andy Stewart

Editorial Director Christopher Holder

Wayne Coyne, lead singer of The Flaming Lips, heads to FOH to listen to his vocal mix.

One of the greatest advantages a live front-of-house engineer has over a studio mix engineer is that, by and large, their clients never get to hear the mixes they create. Sure, one or two band members may occasionally jump off stage, crowd surf their way back to the mix position and catch one song, yelling something inexplicable at them before heading back to the stage, but generally speaking the responsibility for the mix is left to the engineer. He or she knows what to do, and they’re left to perform the task to the best of their ability – on the fly.

or heaven forbid, the A&R guy? In reality, it’s a little bit of all of these. The true art of mixing doesn’t simply involve listening to audio signals and twiddling knobs. Just as importantly, mixing involves listening to the people around you, interpreting their thoughts (and subtext) and reaching a point along a timeline where everyone is happy. Anyone can mix something in isolation for his or her own gratification; that’s like falling off a log. But mixing for an audience, a band, the radio transmitters and yourself all at once – that’s another matter.

The role of the studio mix engineer differs from this in at least three respects. Firstly, a studio mix is based on a repeatable event, initially replayed over and over during the recording session, then over and over again during mixdown, over and over yet again during mastering, and eventually over and over on the radio, in peoples’ homes, the supermarket, at nightclubs and so on. Some mixes are played over and over for decades – becoming iconic, woven into the social fabric like a woven fabric-y thing.

This somewhat belatedly leads me to a question I’ve been pondering for weeks, which first prompted this editorial: who is playing what role in the studio during the mixing process these days? I ask because a trend I’ve noticed of late (and experienced first-hand many times in the last 12 months) is beginning to call into question the traditional role of studio mix engineer.

Clearly this makes one hell of a difference to the way recorded audio is handled – before, during and after the event – which leads us to the second point of difference between live and studio mixing. These days, each sound, every individual performance, and every moment in time can be crafted and honed, manicured and tweaked in ‘unreal-time’. Thanks to our great friend the ‘Save’ button, everything computer related can be recalled and reworked, sometimes interminably until one of three things happens: everyone is satisfied, the computer blows up or someone dies of old age. The same applies to live mix snapshots in some circumstances, but even then the performances themselves are still (mostly) live. Time is no longer linear in the studio, although time is still finite in human terms – something some people seem to have forgotten. All this has a profound effect on the approach and overall outcome. Which leads us to the third difference (there are others but we’ll restrict it here to three for brevity’s sake). With studio mixing today, the artist also tends to be the mix engineer. Now, this particular point, perhaps like no other, tends to send a shiver down the spine of live engineers. I’ve personally witnessed one drop his JD & Coke and stare transfixed at the wall upon hearing this grievous concept, conjuring in his mind the horrifying scene of a band crowded around him at front-of-house, each member asking to be turned up, made wetter, made brighter, made phatter and so on. Which leads me to the big question that must be asked, but rarely is, about mixing: “For whom are these mixes engineered?” Is the ‘nailed mix’ the one the engineer is unreservedly ecstatic about but the singer dislikes, or the one that sounds good at front-of-house and nowhere else? Is it the one the singer is happiest with perhaps, or the drummer, AT 8

More and more, artists themselves are taking mixing into their own hands in the same way they did with tracking a decade or so ago, trying their best to get their mixes sounding good before getting ‘a professional’ in to ‘finish things off ’. Sometimes the results are great; sometimes so so. In many ways, embarking on a mix is like trying to design and build your own house: while it might ultimately be cheaper to go DIY, and seem fairly straightforward on the surface – “How hard can it be? I’ve got the tools!” – you may just find it’s a lot harder than you think. If, on the other hand, you involve an architect and a professional builder from the get-go, sure it will cost more, but the outcome will almost certainly be better. So it is with mixing. But if, as an artist, you intend to work this way either now or in the near future, by all means go for it. Just do yourself a favour: after you’ve mixed exhaustively for days and weeks, don’t bother taking ‘stems’ of this work to a professional to ‘finish things off ’. If you plan on getting help towards the end of a project, either because you want to save money, keep some artistic control over the result or get a mix credit on a record, don’t bother creating even more work for yourself by making stems. Bring your computer, or at least the mix session files, with you to the mix engineer’s workplace, and present the multitrack file to them as you left it. (Discuss the ins and outs of this with them first of course.) It will save heaps of time in the long run, and the results will be far better. By creating stems you’re only embedding all your mistakes invisibly into new files, and limiting the other person’s capacity to do their job. It’s like building half the walls of a house and then calling in an architect and saying: “design me the perfect house Joe, but whatever you do, don’t touch these walls, they’re priceless!” I’ve done the stem-sell research – I think I prefer the multitrack files.

Online Editor Brad Watts Art Direction & Design Leigh Ericksen Additional Design Dominic Carey Advertising Philip Spencer Accounts Manager JenTemm Circulation Manager Miriam Mulcahy Proof Reading Calum Orr Regular Contributors Martin Walker Rick O’Neil 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 Distribution by: Network Distribution Company. AudioTechnology magazine (ISSN 1440-2432) is published by Alchemedia Publishing Pty Ltd (ABN 34 074 431 628). Contact (Advertising) 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:

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. 27/6/2011.

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He was once the man from Was (Not Was); he was also a producer of The Rolling Stones, Bob Dylan, Bonnie Raitt, Randy Newman, Willie Nelson, Joe Cocker, and many others. Now he just is. AT caught up with the cat in the hat. Mark Moffatt is an ex-pat Australian who’s engineered his way around the world for much of his life. With an impressive CV and a string of well-known hits under his belt, including I’m Stranded by The Saints and Treaty by Yothu Yindi, these days Mark is one of Nashville’s most prominent producer/engineers.

Readers Letters. News and new product information. Megastick Fanfare are the subject of this issue’s psychological profile. According to the band’s mix engineer they should be given a wide berth by anyone who’s contemplating working with them in the future! Around the studio traps, featuring Pony Music and Rancom St Studios. Martin Walker is discovering hidden latency in Firewire interfaces Brad Watts is urging us to tighten up over the winter months. Spring should sprout new MacPros. Rick casts a net or three into the murky waters of the audio industry to catch a paying client or two.

Stav is back, and this issue makes light work of a tricky session with preparation... and MS technique, which makes all the (sum &) difference. In this two-part exploration of stereo panning techniques, Andy Stewart explores how panning helps recreate three-dimensional space, and maintain balance. This issue Rob Squire reveals the component most likely to fail in your pro audio equipment – the capacitor. Why do they ‘dry out’ and is it anyone’s fault?




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YOUR WORD Readers’ Letters


Gidday Mark, Just wanted to write to you and say that I enjoyed your review of the Heil PR35 mic – very informative. I have some questions about the mic’s potential use in field recordings too, if I may. You talk about the mic’s higher-than-normal output. I found this most interesting because I record a lot of Aboriginal singers, many of whom sing at lower-than-normal volumes. In fact, on a recent trip south, I bought two Rode D-Power modules, which provide an extra 20dB of gain while adding only a minimal amount of noise. I’ve even bought a couple of condenser hypercardioids to get a manageable signal from people whose personal space feels invaded by normal microphone placement. Gradually I’m assembling a case full of mics for most situations, so I read your review with great interest. My first question is: would the PR35 offer any volume advantage over my Sennheiser 521s, or would it fall into the ‘con’ you mention of “not being too great for quiet, delicate acts” where the mids are accentuated? Secondly, you mentioned the PR35’s off-axis rejection of noise, and in your review your emphasis was on the way this characteristic behaves in relation to spill from wedges, but could it also reject community noise: dogs, cars, shouting, wind-rattles and howling, etc.? Could it also reduce reflections from around the room (since we often record in less-than-ideal surroundings)? Finally, one last question. Some of the things you list as ‘cons’ shouldn’t affect my circumstances: handling noise (we use mic stands and suspension mounts), susceptibility to plosives (Aboriginal languages don’t have aspirated plosives, so even when our singers sing in English, they don’t accentuate Ps, Ts and Ks that much. Does that sound about right? It looks like the PR35 would be a good mic to handle live situations too (and we do some concerts), but if you can answer these questions I’d really appreciate it. Thanks. Alan Rogers. Humpty Doo, NT. Mark Woods Responds: G’day Alan – thanks for your letter. It sounds like you do some interesting recording up there and I suspect the Heil Sound PR35 would work well in your situation. AT 14

In terms of output level, the PR35 will have noticeably more level than your Sennheiser 521s, but it’s still lower than many condenser mics so you may still need your gain modules if you need to record a fair way away from the source. I set up a test with a Heil PR35 and a small diaphragm condenser with a hyper-cardioid capsule after I got your letter, and recorded a guitar/vocal from increasing distances, up to 10 metres or so. The Heil needed around 10dB more gain than the condenser, and the preamp became noisy earlier at long distances (over five metres.) Tonally the Heil is mids-forward and articulate, so soft, delicate vocals will be picked up clearly, and the words should be easy to understand, but there is a risk it may also be hard sounding on some sources. The off-axis rejection and focus was better than the condenser and it would definitely help reduce background noise. Wind noise was much lower through the Heil mic too, especially with the built-in two-position high-pass filter engaged. It really does hone in on the subject and will reduce room reflections nicely. Its ability to focus was maintained over all distances. Although it’s been designed to be handheld or used close to the source, I was impressed by the Heil’s tight directional control and clear articulation even at long distances from the source. I’m also confident it will work well in live concert situations, especially for performers who don’t project strongly, with the qualification that they stay fairly closely in front of the mic. Hope this helps! Mark Woods. –– FLAME CHASED & LICKED

Dear Stav I’m in Melbourne and do some recording of vocals (amateur and semi-pro singers) from time to time. Having read Mixing with your Mind and liking the idea of ‘Chasing the Flame’, I’ve found a different implementation of your concept. I record two tracks at a time – one channel from a static large-diaphragm condenser mic (Rode NTK) and the second channel of the same performance with a handheld Beyerdynamic M88. The results have been amazing in terms of capturing the performance quality and the live dynamic feel of the vocals. I know there are probably phase issues and all manner of variations between the two channels (which I

pan hard L&R in the final mix by the way). Yet I like the result from having the vocalist ‘chase their own flame’ in the studio. Or is it just that, with a vocal mic in hand, these performers give more life to their renditions in the studio? I offer this approach for others to try (or critique). Thanks for your AT columns – they are inspiring and useful! Steve Smith. Melbourne. Stav Responds: Congratulations Steve on your fearless approach to recording a vocal! Despite the unpredictable phase and delay-related artifacts of your approach it seems your sense for what might work has served you well. This is exactly the kind of wild experimentation we need more of. It’s easy to imagine why things shouldn’t work but when magic happens it’s equally easy to construct a theory that explains how it worked. The important thing is to keep bending the rules to see which ones break to unveil original techniques that, in hindsight, become simple approaches that work a treat. Clearly the reason this worked out so well was your quality of movement and sense of timing and distance. I’ve not heard or tried this particular combination myself. If the music track is densely populated with lots of instruments you may not have the space to appreciate it, but I can imagine it working as a contrasting effect. Good mic technique by the singer is always great, but when I chase the flame for an artist I do take it to extremes (once I nail the timing) with movement in three dimensions, not just distance, but height, angle, and on and off polar pattern axis as well, to give extra colour to the performance. It’s a buzz – looks crazy – but sounds amazing when you get it. Well done! Stav. –– RE: ED SPACE ISSUE 81

Bula Andy! [Bula means ‘hello!’ in Fijian – Ed] I’ve just returned from my fourth visit to Fiji where I took my laptop and mobile preamp, 240 and 9V mixers, a Sony Walkman MD, multiple leads and mics... and my newly acquired Zoom H1 (bought after reading the review in a recent edition of AT).

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I had a special job to do on Yasawa Island where I used the laptop, but the H1 got a hell of a workout (filled two 8GB cards over the two-week stay). I used it in the Vunaqoru Village way up the Sigatoka River, the Blue Lagoon Resort on Nacula Island, the Gateway at Nadi Airport, Robinson Crusoe Island, the Hideaway Resort, and at a Church Service near the Hideaway. It was excellent.


I mostly used the ‘auto function’ due to the difficulty of getting the musos to stop and start while setting levels. The auto function also works well while you’re following the sound along a beach etc. Next visit I’ll be taking the laptop and preamp, the MD for backup of important files (I’ve got Vista!), and the Zoom H1 with plenty of SD cards. The rest will stay at home… “Great idea,” said the wife! Great article and sound Andy. I too have become a bit of a missionary in relation to strings, plectrums etc. One thing... the three strings on the uke is not an uncommon site in Fiji. I have seen many like that… “easier to play” I’ve been told. I’ve even bought a uke myself along with a chord chart. You may end up recording me on a beach somewhere in Fiji!

The new SoundField SPS200 ‘software controlled’ microphone generates mono, stereo, 5.1 and beyond and gives the user complete control over all microphone parameters within the audio workstation. Designed and built to the same exacting high standards as other SoundField models, the SPS200 is supplied with its own Surround Zone software for both Pro Tools HD and VST platforms.

Vinaka! Brian Giblin, VIC. Andy Stewart responds: Bula Brian, and cheers for the email – small world! Amazing to hear how many times you’ve been back to Fiji. Not sure if you’ve been driving your wife crazy by taking all that recording gear with you each and every time you go there; sounds like traveling lighter next time will probably suit you both a little better! I wish I’d been more confident about the ‘auto gain’ function on my Zoom recorder when I recorded over there recently. The final recording levels were perilously low in the end – not good. ‘Never trust an auto function’ is my general motto with respect to new gear, or old gear for that matter, especially on something so inexpensive. But perhaps I should reassess this reservation. My logic has always been: if they can’t make a decent compressor for 300 bucks, how can they make an auto function work properly for $1.45? I gave it another go at a Snoop Dogg concert recently though and it seemed to work okay. Wasn’t sure if the artefacts I could hear were due to the auto function or the fact that the unit was getting pulverised by 117dB.

By using a single SPS200 microphone and laptop computer, making world class surround and stereo recordings has never been so easy.

Thanks again for the email Brian. Maybe we’ll cross paths over there some day. Send me a song file – I’d love to hear it. Andy Stewart.

Studio Connections Australia Pty Ltd


11/ 41 - 49 Norcal Road Nunawading VIC 3131 Tel: 03 9874 7222 Web:


A few months ago I had some ongoing issues with some rascally rodents invading my studio space. Thanks to some rat whisperer advice from AT regular Calum Orr, I managed to eradicate the ‘issue’ with extreme prejudice. Problem solved. Then yesterday I plonked myself down to patch in this month’s studio indulgence and found that the dearly departed squeakers had in fact set up camp amongst the patch leads of my preamp rack. The telltale ratty tic tacs were everywhere, not to mention acorn husks and other miscellany. Thankfully all the leads were un-nibbled but it was a timely reminder that studios are generally full of warm nooks and hollows and it’s worth keeping your eyes and ears open for more than just the rustle in your tweeters. I’m off to check for signs of life... Basil! Andrew Bencina AT contributer/Winterhill Music


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All contributers to Your Word this issue will receive free copies of debut albums by Owls Of the Swamp (Go with River) and Megastick Fanfare.

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Shure SM57/Copy

IN BRIEF COMMUNITY RADIO GOES DIGITAL Community radio stations in Melbourne, Sydney, Adelaide, Brisbane and Perth recently commenced digital radio services, providing communities in each city with a new way to listen to the most diverse, accessible and independent media sector. The national launch for the historic event was held at one of the oldest community stations in the country – 3RRR FM in Melbourne. It’s a significant moment in the development of community radio. Adrian Basso, President of the Community Broadcasting Association of Australia, emphasised the importance of the occasion, saying: “Every week, millions of Australians tune into community radio to enjoy information and entertainment that speaks directly to their heart and their community. After years of lobbying, it’s a great moment to witness the first community stations enter the digital radio age.”

CALL THE DOCTOR Empirical Labs is stoked to announce the DocDerr, a 500-series format multi-purpose channel strip. The ‘tone enhancement module’ (ELI’s term) houses six sections of digitally controlled analogue processing including a low-noise preamp, followed by four sections of EQ that include one selectable high-pass and three parametric bands. The dynamics section provides renowned Empirical Labs compression, as well as a tape emulation circuit to soften high frequencies and clipping. A Mix knob allows the user to blend the uncompressed EQ’s signal with the compressed and saturated signal. A ‘Bad!’ hard clip indicator monitors most sections for internal clipping. The DocDerr model EL-Rx will be available with either horizontal or vertically oriented panel configurations, and begins shipping July 2011.


$999 |

Well, this got our attention. Coming soon is the Digitech iPB-10 programmable pedalboard. Incorporating the power of the iPad, the iPB-10 combines the simplicity of a pedalboard with the flexibility of a multi-effects unit. The Frankenstein pedal allows you to create your ultimate pedalboard on your iPad: designed by simply dragging and dropping up to 10 pedals to each pedalboard, then adding an amp and cabinet to each setup. With 87 different pedals, 54 amps, and 26 cabinets, the options are truly vast. Traditional multi-effects operation then gives you the flexibility to change the entire configuration of your signal chain with a single stomp. The iPB-Nexus app uses the iPad to build and control your pedalboard. Add, arrange, and adjust your pedals with a few finger gestures, and end up with a real-time view of your set-up. Just don’t put your foot through the iPad! CMI (03) 9315 2244 or

Atlantis Asia: (03) 9818 7778 or

PHATTEN UP Moog Music recently announced Phatty OS v3.1, a firmware upgrade to the Phatty family of analogue synthesisers. New features include Volume Velocity Sensitivity (the ability to map note velocity to output volume), Legato Glide, LFO Reset and Sustain/Hold Mode. There’s also provision for 14-bit MIDI: Phatty synths can now send and receive high-resolution 14-bit MIDI messages for critical voltage-controlled parameters. There’s also MIDI output filtering (to control which MIDI CC messages are sent from the Phatty), expanded MIDI clock sync options (22 clock divisions including triplet values for both LFO and Arpeggiator MIDI sync – clock divisions range from 32nd triplets up to four bars), along with vast improvements to the arpeggiator. Available free from www. Audio Chocolate: (03) 9813 5877 or


Audix’s Micro-D miniature rim-mounted drum mic encouraged demand for a similar product for hand percussion. Enter the Micro-HP, which utilises a cardioid polar pattern to capture the attack of the head as well as the tonality and transients of percussion instruments. A rubber shock mount and protective anodised aluminium ring provide vibration control, while the gooseneck design allows placement of the Micro-HP into virtually any position. The ‘mini’ design keeps the mic out of the way of the player’s hands and the mounting system eliminates the need for a mic stand. Available in matte black, the Micro-HP includes a two-metre integrated cable and phantom power adapter, windscreen, and DCLAMP-Micro for lug-nut mounting. PRODUCTION AUDIO SERVICES (03) 9264 8000 or

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Records Shitty Album



The first thing Brad Watts did when he and his family moved into their new house in central Victoria recently was setup a makeshift pyre in the backyard. Made up of leftover bricks from the house construction, Brad proceeded to burn a diverse array of audio-related products: AudioTechnology and Tape Op magazines, even a couple of Akai Samplers from last century. Call it a purging, call it a protest, call it what you will – one thing’s for sure, it was an environmental hazard. In fact, it looks like some of the so-called combustible material would have had a hard time igniting if it had been thrown into an active volcano! Nice one Brad. Haven’t you heard of eBay?

JZ GOES C12 JZ Microphones is now selling its LDC Vintage 12 condenser microphone, a version of the legendary AKG C12. Following on from its successful Vintage 47 and Vintage 67 microphones, inspired by the Neumann originals, JZ Microphones’ Vintage 12 incorporates and enhances qualities of the original with features unique to JZ, such as the ‘Golden Drop Technology’, which reportedly provides a more detailed and transparent capsule response. The replacement of the original C12 tubes with high-quality transistor technology provides equivalent frequency response with fewer maintenance and care issues.



$1100 |

The Nagra SD hand-held solid-state audio recorder is now available in Australia. The ultra-portable rugged little tyke uses removable SD memory cards, records PCM mono or stereo WAV files at frequencies of 24kHz up to 96kHz at 24-bit, or with MPEG Layer II / MPEG III compression, and is powered by two ‘AA’ batteries providing in excess of 10 hours operation. The Nagra SD offers instant start up, one-button recording, automatic level control, an internal loudspeaker and a low-cut filter. A set of switches on the rear of the recorder provides the user swift access to important functions without the need to enter menus. The Nagra SD is USB 2.0 compatible and no additional software or drivers are required to access files via a PC or Mac. BROADCAST WORKSHOP (03) 9329 7655 or


Moog Music has announced the release of the MF-108M Cluster Flux. The bawdily named Cluster Flux is an ultra-flexible sound-sculpting tool for studio or live use. More powerful than traditional chorus/flange effects, the Cluster Flux adds modulation sources of multiple LFO waveforms: sine, triangle, square, saw, ramp and random, to create a wide variety of modulated delay effects. MIDI in allows control of delay time, range, feedback, output level, LFO waveform, LFO rate, LFO amount and mix. Additional features include the ability to sync LFO modulation effects to MIDI Clock or Tap Tempo, control delay time via MIDI notes, CV control of delay time, LFO rate, feedback, LFO amount, and mix. A second output is configurable via DIP switches for different types of stereo output. AUDIO CHOCOLATE (03) 9813 5877 or

Solid State Logic recently announced the release of V2 Firmware for the Delta-Link MADI HD. The free V2 Firmware update supports ProTools Native cards and introduces support for longer Digilink cables between Delta-Link and ProTools Native, ProTools Core or ProTools Accel cards. The Delta-Link MADI HD is a professional MADI-to-ProTools HD converter for studio, live and broadcast applications, providing a simple bi-directional connection via two ProTools HD Ports and two MADI optical connections. Amber Technology: 1800 251 367 or

GOODBYE EARTH Neolev magnetic levitation dampers are the latest development from Triton Audio. Placed under studio monitors, CD/DVD players, or turntables, the levitation dampers ‘eliminate acoustic feedback’ and ‘mechanical distortion’ to bring out the full potential of your gear. Based on two repelling Neodymium magnets, one single NeoLev lifts up to 8kg. Three or four will set your nearfields hovering – seriously – completely disenfranchised from Planet Earth. NeoLevs are also adjustable to ensure perfect levelling. Cancellation of contact induced resonance is apparently effective in sound improvement as forces perpendicular to the device are stopped from being transferred. We’re giving some a run right now, and will let you know the outcome. Professional Audio Services: (02) 6059 1652 or

ROCKIN’ IN THE FEE WORLD Tom Cruise has gone and done it again. This time he’s decided to rock out as fictional ’80s heavy-metal character, Stacee Jaxx, in a film adaption of the 2006 Broadway musical, Rock Of Ages. Early news reports and pictures have seemed more concerned about ‘Stacee’s’ leather pants, fake tattoos and beefed up bod, but can he sing? According to his fawning co-stars he can, but other (similarly ficticious) reports have suggested a whole new Waves plug-in had to be developed to cope with his over-the-top rantings. DeCrapwit, as it’s known, reportedly feeds the plug-in’s input directly to a sidechain which feeds the singer’s in-ears so no-one else hears it… the output is a triggered sample of Bon Jovi.

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Vermona VSR-3 Spring Reverb

IN BRIEF HAVE STUDIO: WILL TRAVEL Looking for a recording studio that’s completely portable and compact? Sick of renting studio space? Mark ‘Sparky’ Paltridge has popped up from time to time in the pages of AT with his shipping container-based studios. Mark was weary of setting up studios in rented buildings and eventually decided he’d make his own completely portable by building it into a shipping container. It transpires that Mark has had many a request for portable studios since and has opted to go into business building these beasties for others. Spark One Container Studios provides complete control and tracking rooms that offer professional acoustic standards within a compact and portable environment. Or you could go for a couple of containers, housing your control room in one container and the tracking room in another. Spark One Container Studios: 0416 262 319 or

RADIAL ENGINEERING WORKHORSE The Radial Workhorse WR8 is an eight-slot 500-series mixer that is 100% compatible with all standard 500-series modules. Individual XLR inputs and outputs are complimented with parallel 1/4-inch TRS connectors and ProTools-compatible 25-pin D-Subs. This enables any of the eight module slots to painlessly integrate within the framework of today’s DAW environments. A special feed function also enables modules to be connected in series without having to hard-patch cables. Installing the WM-8 Workhorse eight-channel mixer section can be completed by the user with information supplied on Radial’s website. Amber Technology: 1800 251 367 or

MONSTER MASH The Monster Compressor is a FET based audio unit that’s produced in Poland by Looptrotter Audio Engineering. It’s of discrete construction and is devilishly accurate and frighteningly musical all at the same time apparently. It isn’t a clone, but rather hangs refreshingly by its own rack ears in an increasingly bewildering audio market. Being a monster it’s got every feature you could possibly ask for in a dynamics processor, including parallel compression and tube saturation processing. It might be yellow, but one thing’s for sure, you can read the settings from a mile away if the wind is blowing towards you. Despite its apparent complexity the Monster Compressor is also reportedly incredibly easy to use. We’d print the specs but they’re in Polish. Available at an introductory price of $3695. Mixmasters: (08) 8211 6211 or


The ProFX22 will slot into budget-conscious medium to large venues that lack space or funds for swags of outboard. To compensate for the ‘missing rack’, tools like compression and graphic EQ are included in the console. The ProFX22 has everything needed to make your gig happen and all the necessary channels to record a large band. Plus, USB I/O lets you record the show or stream music straight from a laptop with the included copy of Tracktion 3 Music Production Software – other DAWs are also supported too of course. The ProFX22 is loaded with 16 Mackie low-noise mic preamps, three-band EQ, integrated effects, and precision seven-band graphic EQ – all housed in the traditional ultra-tough enclosure you’d expect from Mackie, with additional impact-resistant side-cheeks. MUSIC LINK (03) 9765 6565 or


TC Electronic has put together a ‘power monitoring’ bundle: the PLM3A as it’s known. The monitoring system is the combination of three state-of-theart elements that form what TC describes as ‘the ultimate’ audio monitoring chain. First link in this chain is a set of Dynaudio Acoustics M3A threeway monitors, which are powered by the second link: a Lab.gruppen PLM 10000Q amplifier, which features an impressive 2300W of power per channel at 4Ω . Last but not least, Lake and TC Electronic processing comprises the third, ensuring optimisation of the system with any room for a quite spankingly impressive listening outcome. The M3A speakers feature a pair of 300mm bass drivers, complemented by two 150mm piston midrange drivers with massive 75mm coils. AMBER TECHNOLOGY 1800 251367 or

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

Visit the Soundcorp Showroom and find The Sweet Spot, a purpose-built listening room designed so you can compare the latest PA speakers and studio monitors. Best Brands. Best Choice. Best Price. Soundcorp.

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 21



Filther AU Filter Plug-In

IN BRIEF MOTU HEARS THE STORM MOTU recently announced it’s working closely with Apple and Intel to develop future products that take advantage of Thunderbolt, the super high-bandwidth, standard deployed by Apple on its latest generation MacBook Pro and iMac computers. “MOTU has often been the first to adopt new connectivity standards, such as Firewire, in our video and audio I/O products,” said Jim Cooper, MOTU Director of Marketing. “We’re excited about Thunderbolt because it clearly represents the future of professional I/O. MOTU’s strong partnership with Apple and Intel lays the groundwork for exciting new possibilities for Thunderbolt-equipped MOTU products in the near future.” Major Music: 1300 306 670 or

WAVES MODELS MORE Developed in association with producer/engineer Eddie Kramer, the Waves’ MPX Master Tape is modelled on a rare machine consisting of an Ampex 350 transport and 351 electronics used by Kramer at Olympic Studios in London on his recordings of Led Zeppelin, The Rolling Stones and Jimi Hendrix. With adjustable tape speed, bias, flux, and wow and flutter, the MPX provides comprehensive control over the contours of any sound. The Ampex 350 and 351 valve tape machines were mainstays of the recording industry and were used to record literally thousands of hit albums.


Celemony Software has revealed a new music restoration program, Capstan, which removes wow and flutter from musical material, and will be a boon for professional restoration and mastering studios the world over. For decades, music has been recorded on mechanical storage mediums, and for over 100 years these recordings have been plagued by the same problems: wow and flutter caused by the mechanical transports that drove them, and the mediums themselves. Capstan will remove such indiscretions – whether on tape, cassette, vinyl, wax or shellac. The medium is of no relevance – Capstan works even if the material has been copied several times or digitised at low resolutions.

Sound & Music: (03) 9555 8081 or

Capstan is hardware-independent, and runs as a stand-alone application under 64-bit operating systems (Windows or Mac OS). Due for release in mid June for the grand old sum of €3790 !


ELECTRIC FACTORY (03) 9474 1000 or

Vintage Compressors from Native Instruments was created in collaboration with Softube. The VC 76, VC 2A and VC 160 are painstakingly accurate emulations of three of the most popular compressors in studio history. Each has its own flavour, with the plug-ins adding the convenience of software side-chaining and parallel compression. The VC 76 recreates UREI 1176-styled compression. The VC 2A is an emulation of the ubiquitous LA-2A, while the VC 160 is based on the dbx VCA compressor. They’re available for download only, either individually for US$119, or as a three-pack for US$229. CMI: (03) 9315 2244 or

THE LION SLEEPS Apple recently announced that Mac OS X Lion, the eighth major release of the operating system, will be available in July as a download from the Mac App Store. Some of the stupendous features in Lion include new Multi-Touch gestures, system-wide support for full-screen apps, and Mission Control (another crack at viewing everything running on your Mac at once). The Mac App Store will also be integrated with the OS, and a completely redesigned Mail application – excellent. Download the 4GB update for $31.99. Apple:

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New from Waves is a somewhat esoteric plug-in that models three microphones used exclusively by the royal family for speeches on momentous occasions. These original three priceless microphones date back to the 1920s and ’30s, and were tuned especially for its specific users: King George V, King George VI, and Queen Elizabeth. Beautifully designed and decorated with gold, silver, and chrome adornments bearing the royal coats of arms, each is a one-of-a-kind masterpiece. Well… thankfully The King’s Microphones plug-in offers none of that guff, but it does provide the sonic character of each mic. We’re not entirely sure who’d be keen on such a plug-in, but top marks for the crossmarketing; vis-à-vis The King’s Speech. SOUND & MUSIC (03) 9555 8081 or


iSyn Poly For iPad



$169 |

Punch, from Rob Papen, is a speaker busting, trouser flappin’ software instrument of the ‘rhythm programmer’ persuasion. Delivering synthesised drums with a stunning array of features for the contemporary electronic producer, Punch provides a choice of synthesis and built-in samples with which to build your own unique sounding drum kit. It also allows you to load your own samples to add to the onslaught, and includes built-in sequencers, allowing you to have multiple patterns at your fingertips. These grooves can be triggered in a live environment to build a song, and are also great for improvisation and inspirational jamming – Mac or PC: AU, VST, and RTAS in 32- and 64-bit flavours.

SONNOX FRAUNHOFER PRO-CODEC Sonnox has come up with a terrific new tool, the Sonnox Fraunhofer Pro-Codec plug-in, which allows you to audition various audio codecs before you commit to a bounce or mixdown. Until now it’s been impossible to audition and then encode multiple formats in real-time within a DAW. The Sonnox Fraunhofer Pro-Codec makes this possible, and includes extensive monitoring tools and built-in encoding/decoding features. Supported codecs include mp3, mp3 Surround, AAC-LC, HE-AAC and HE-AAC v2, mp3-HD and HD-AAC (lossless codecs). Supported platforms include Avid ProTools 7, 8 (RTAS and M-Powered), ProTools 9 (RTAS) and ProTools HD|Native, VST and Audio Units – Intel OSX 10.5 or later and Windows XP/7. Audio Chocolate: (03) 9813 5877 or


AUDIO CHOCOLATE (03) 9813 5877 or


$139 and $199 |

Recently bolting from the Roland stable are two USB interfaces: the Duo-Capture and Tri-Capture USB audio interfaces. The Duo is equipped with two audio inputs and two outputs, and allows side-by-side musician/engineer sessions with no additional mixing hardware. The unit includes dedicated controls for input and output volume, and all-important direct monitoring. The Tri-Capture offers a mixer-like design and dedicated front-panel controls, along with 24-bit/96kHz operation: plug a mic into the XLR input, an instrument into the 1/4inch jack, and a stereo device into the aux input. The unique ‘Rec Mode’ section provides instant selection of recording scenarios, including ‘Loop Back’ for web streaming applications. Both are USB powered and include Cakewalk Sonar X1 LE. ROLAND CORPORATION (02) 9982 8266 or


Steinberg has announced Halion 4 is now shipping! Combining sampling and synthesis technologies, mixing options, its unique morphing filter and FlexPhraser arpeggiator, the enhanced MegaTrig tool, more than 40 studio-grade effects and a library spanning over 1600 sounds, Halion 4 is quite a prodigious virtual instrument. Halion’s most prominent advantage is arguably the implementation of VST 3.5, with its innovative Note Expression for editing expression data polyphonically. Comprising both 32-bit and native 64-bit versions for Windows 7 and Mac OS X 10.6, Halion 4 runs as a standalone version or as plug-in within all VST and AU hosts. Yamaha Music Australia: 1800 805 413 or

SOLINA? Loomer String is a virtual instrument that emulates the timeless sound of the string synthesiser. With optimised audio algorithms that won’t overload your processor, and offering rock-solid performance and integration with any MIDI hardware controller, String is particularly well suited to live work. Offering an intuitive interface and enough parameters to sink a cello, String looks capable of covering string-synth bases. Available as AU, VST, and RTAS plug-ins, and standalone via OS X, Windows XP and – wait for it – Linux. Price: 50 quid. Loomer:

The Avid Scorch app transforms your iPad mobile device into an interactive music stand, score library, and sheet music store. Just the ticket for viewing beautiful scores anywhere. Scorch generates true, interactive notation utilising the award-winning Sibelius engine and makes it easy to adapt scores to your needs. Transpose the music to play in a key that’s more comfortable for you or your vocalist, change instruments, and convert to and from guitar tab. Alternatively, focus solely on your instrument and remove other parts to eliminate distraction, then, when you’re ready to perform, engage ‘Music Stand’ mode to turn pages quickly and stay in the moment. Grab it at the App Store.



Audiofile Engineering:

Audiofile Engineering has acquired Quiztones, an ear training application for audio professionals and musicians, and released it as an App for the iPhone, iPod Touch, and iPad. Quiztones is designed to train your hearing to recognise frequencies – a valuable skill for tasks like mixing, adjusting EQ settings, and identifying feedback. The application uses various quizzes to engage users in training exercises from simple frequency tones and pink noise, to real-world sounds like frequency-altered drums, guitars, bass and vocals. Quiztones requires iOS 4.0 or later on an iPhone, iPod Touch, or iPad and is available through iTunes for $3.99.

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AKG WMS 470 wireless system

KLA FROM QSC $3999 |


Built on the same power platform as the K and KW Series powered loudspeakers, QSC has launched KLA – an easy to deploy, easy to tune line array. The KLA system is comprised of two models: the KLA12 12-inch, two-way, line array element and the KLA181 18-inch subwoofer enclosure. Both models feature a 1000W Class-D power module and a self-contained rigging system that enables users to quickly assemble and disassemble the system. The fixed vertical splay angle of each box is 18°, allowing 90° arrays to be configured using only five boxes. KLA is fully supported in EASE and EASE Focus and is available in black and white finishes.

MIDAS & KLARK TEKNIK MANAGEMENT RESTRUCTURE Organisational changes at Midas and Klark Teknik have seen the departure of former MD, John Oakley. Both firms are now undergoing a “restructuring of the management team” apparently to accommodate Oakley’s departure and allow for better integration within the shared services of Uli Behringer’s Music Group. Newcomer Ian Riley, who joined Midas and Klark Teknik from Harman earlier this year, takes on the role of VP of Operations and R&D, while Jonathan Chitty is VP of Customer Support. On the manufacturing side, Music Group has invested $10m in a dedicated manufacturing facility for the Midas and Klark Teknik brands. Alex Cooper remains as R&D consultant to Midas and Klark Teknik to oversee ongoing and future products. Uli Behringer, Music Group CEO, said: “I am sad to see John leave the company after all he has done for Midas and Klark Teknik. We wish him all the best for the future. With the strong team we have in place now, Midas and Klark Teknik is already seeing a strong increase in revenue streams and the volume of products shipped.” All this says very little about why John Oakley left. Watch this space next issue for an interview with John about his recent departure and future plans.

MORE WIRELESS Shure has revealed two products to compliment the UHF-R series of wireless systems. The UR3 is a plug-on wireless transmitter designed to work with the UHF-R series, which will turn virtually any type of wired microphone into a wireless microphone. Unlike other plug-on transmitters, the UR3’s design allows users’ hands to cover signal and battery indication LEDs without the light emitted being picked up on camera. The UR5 is a portable camera mount single-channel receiver for UHF-R wireless transmitters. The receiver (roughly the size of a belt-pack) includes diversity antennas, full bandwidth scan and sync at the touch of a button, headphone monitoring output jack, mic and aux-level balanced output.

TECHNICAL AUDIO GROUP (02) 9519 0900 or


As a successor to Outline’s H.A.R.D. series of self-powered floor monitors, the iSM series, featuring the iSM 112, iSM 115 and iSM 212, delivers higher SPL and full remote control of speaker parameters through Outline’s iMode Intelligent Digital Loudspeaker Control Platform. The iSM Series is designed for fixed or touring floor monitoring applications, but is especially useful in TV production where a super-low profile cabinet is advantageous. Each cabinet features Outline’s iMode technology for parameter control via a web browser and LED arrays on both sides of each cabinet for visual status of parameters. All cabinets are fitted with eight mounting points for wheels or flying rings. AMBIENT TECHNOLOGY (03) 9689 1777 or

Jands: (02) 9582 0909 or

YAMAHA REDUCES FEEDBACK WITH RND As part of a new partnership with Rupert Neve Designs, Yamaha Commercial Audio Systems has announced the Rupert Neve Designs 5045 – a two-channel analogue device designed to reduce the tendency of a system to feedback. The unit enables up to 16dB of additional gain before feedback when conditions allow. Easy to use, the device reduces background sounds, effectively enhancing the main audio source. The RND 5045 uses transformer-coupled inputs and outputs that produce the same classic sound Rupert Neve products are known for. Available during the third quarter of 2011. Yamaha Music Australia: 1800 331 130 or


Powersoft has emerged with some new M Series DSP+ETH amplifiers. The M Series DSP+ETH amps feature 24-bit/48kHz A/D and D/A converters with 56-bit internal processing. Each amplifier is housed in a single-rack unit, rugged metal case weighing in at modest 7.4kg. Powersoft’s bridgeable switch-mode fixed frequency Class-D outputs provide cooler circuits for stable performances over time for a longer amplifier life and reduced operational costs from power draw and HVAC cooling. The 20000DP is sold packaged in a touring rack, which includes three amps, distro and breakout. For a stand-alone version, check out the Lab. gruppen PLM20000Q. PRODUCTION AUDIO SERVICES (03) 9264 8000 or

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Inspired by the past Built for the future




Premium quality keyboard

Ergonomic design

High level of control

Full connectivity

Superb audio quality

Powerful analog sounds

- High quality semi-weighted 61-keys keyboard, with velocity response and channel aftertouch - New exclusive duophonic aftertouch

- 2 audio ins,10 audio outs. - Digital Audio: SPDIF out + USB 2.0 - Built-in expression pedal and footswitch control inputs

- Self adjustable front panel, from 0° to 135°, giving you total setup flexibility. - 5.2” TFT screen for clear visual feedback. - Easily transportable cabinet offering a complete mobile solution

- Analog Devices Tiger Shark® processors - TAE® engine - Up to 32 voices of polyphony for a typical patch - High quality effects: Phaser, Chorus, Delay, Distortion, Parametric EQ, Rotary...

- Ultra sensitive 40cm ribbon controller - Modulation and pitch-bend wheels, 3-mode Joystick, 21 potentiometers, 33 rotary encoders, 81 switches

- More than 600 presets created by talented musicians and synthesizer specialists - Create your own patch by connecting independent modules from: Minimoog, ARP 2600, CS-80, Jupiter-8, Moog Modular, Prophet VS

Visit to find a stockist.

Distributed in Australia by CMI Music & Audio 03 9315 2244

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EAW DEBUTS VF SERIES EAW has unveiled its VF Series family of passive products, consisting of five models of VFR two-way full-range loudspeakers: two VFS subwoofers and three VSM stage monitors. Described as “extremely versatile,” the VF Series is intended for both portable and install applications and environments, including auditoriums, theatres, night clubs and DJ setups, bars, and music performance spaces. The VF Series can be considered an update to EAW’s successful VR and FR series, but with enhanced transducer technology and performance. A number of configuration possibilities exist with the VFR and VFS Series, including clusters and arrays, matching speaker/subwoofer pairs or stand-alone units.

Shure has a new flagship series of wireless and has entirely rethought how RF is managed, especially in situations where spectrum is limited or when more than a dozen or so channels are required. It’s called Axient and it’s quite something to witness in action – as AT did in the US recently. The standout feature is Axient’s ability to detect interference and automatically avoid it. Currently, when unexpected RF interference arises, an engineer is either stuck with dropouts or forced to run a backup mic out to the performer. Meanwhile, Axient’s approach provides multiple levels of redundancy. Here’s how:

Production Audio Services: (03) 9264 8000 or

A CROWNING FROM CROWN Crown recently released updates to its popular XTi series of amplifiers, now called the XTi 2 series and comprising four models: XTi1002, XTi2002, XTi4002, XTi6002. So what’s new? Well, to start with there’s an improved front panel, but it’s what’s changed under the bonnet that matters most. A new Harman proprietary DSP engine runs twice as fast as the previous generation, and users now have full control over the new Peak Plus limiter. Attack and release times can be altered as per requirements, along with manipulation of threshold (including the ability to alter it’s crossover point in terms of volts or dBFS). Also new in the DSP engine is a fully featured sub-harmonic synth, with maximum harmonic frequency and crossover controls. Under the Advanced tab is an AC Line voltage monitor, plus easier access to internal PSU temperature monitoring and fan control modes. Jands: (02) 9582 0909 or

AUDIO-TECHNICA BUNDLES UP The BP893cW MicroEarset Omnidirectional Condenser Headworn Microphone is now available packaged with Audio-Technica’s 2000 and 3000 Series wireless systems. The product bundles, otherwise known as ATW-2193a 2000 Series and ATW-3193b 3000 Series wireless systems, come with BP893cW MicroEarset (with black or beige microphone), 2000 or 3000 Series receiver and a UniPak body-pack transmitter. With its ergonomically molded earpiece and unobtrusive one-inch boom, the BP893cW offers inconspicuous placement, and its lightweight low-profile design make it ideal for use in theatrical performances and broadcast studios. The BP893cW includes a 1.4m attached cable, terminated with a locking four-pin connector for use with Audio-Technica UniPak transmitters. Technical Audio Group: (02) 9519 0900 or

• Interference Detection and Avoidance detects RF interference and enables the system to move to a clear and compatible frequency in milliseconds, making even major interference undetectable. • Frequency Diversity transmits full-bandwidth audio on two separate frequencies to ensure seamless, uninterrupted audio for mission-critical channels, even in the face of direct RF interference. • ShowLink remote control enables the user to make real-time remote adjustments of transmitter settings like audio gain from the receiver or a laptop while the microphone is live. ShowLink remote control extends throughout the transmitter’s RF range. • The Axient Spectrum Manager constantly scans the RF environment, performs frequency compatibility calculations to assign clear frequencies to each wireless transmitter. It also continuously monitors, ranks, and deploys backup frequencies automatically. • Smart lithium-ion rechargeable battery packs have zero memory effect, and deliver accurate battery life metering in hours and minutes, time remaining to full charge, and battery health. • Wireless Workbench 6 is a completely new software interface that enables the Axient user to monitor and control the entire system. If you own existing Shure U-Series inventory, you’ll be pleased to hear you can mix and match Axient into your rig. You might want a couple of channels for the main vocals, for example.. JANDS (02) 9582 0909 or


In a joint technical initiative with Audinate and Lab.gruppen (who seem to be everywhere of late), Turbosound has announced the release of the 20000DP amplifier for its FlashLine system. The 20000DP provides the convenience of Lake DSP combined with the power of the Lab.gruppen PLM Powered Loudspeaker Management system. The 20000DP provides four discrete output channels, each rated at 4400W at 4Ω . The 20000DP also contains full-featured Lake Processor modules, each offering settings for gain, delay, standard and linear phase crossover slopes, equalisation, and limiting. The integrated Danté networked audio distribution brings a proven network standard to concert sound reinforcement with reliable, sample-accurate, low-latency audio distribution over ethernet. HILLS SVL (02) 9647 1411 or

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Freecall 1800 441 440

DN9650 Network Bridge Connecting different audio networks together, which often have differing sample rates and clock domains can be a difficult and expensive process. The DN9650 Network Bridge allows all Midas digital consoles and digital I/O hardware, Klark Teknik DN9696 High Resolution Audio Recorders and many other AES50 devices to connect to many different multichannel digital audio networks simply and reliably. Up to 72 bi-directional channels are supported in 1:1 channel mapping between AES50 and the Network Module Interfaces. Current Module Interfaces include: • MADI (AES10) • Dante • Aviom • CobraNet • EtherSound Clock Synchronisation The DN9650 clock synchronisation scheme is divided into two domains with the AES50 domain and third party network domain separated by the Asynchronous Sample Rate Converter. Network Domain Clock Options Include: • External network clock • On board oscillator (AES50 internal clock) • Network Module onboard clock • Word clock input • Video Black Burst input AT 27 Freecall 1800 441 440


Photo: John Mayer

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He was once the man from Was (Not Was); he was also a producer of The Rolling Stones, Bob Dylan, Bonnie Raitt, Randy Newman, Marianne Faithful, Willie Nelson, Joe Cocker, and many others. Now he just is... the cat in the hat. Text: Paul Tingen

Don Was is one of the most iconic names in the world’s production firmament. He’s the go-to producer for top artists looking for a helping hand in realising inspired, soulful recordings and, in many cases, reviving their careers. Was enjoyed his highest profile as a producer in the ’90s, when he worked with big names such as The Rolling Stones, Bob Dylan, Bonnie Raitt, Randy Newman, Marianne Faithful, Willie Nelson, Joe Cocker, and many others – winning a Grammy Award for Producer of the Year in 1995. For a variety of reasons (see boxes) Was has had a slightly lower producer profile since the turn of the century, but his credits during the last decade have still included big names like The Rolling Stones, Bette Midler, Kris Kristofferson, and The Black Crowes. Don Was’ co-production of Lucinda Williams’ new album, Blessed – the main focus of this article – shows that Was continues to operate at the highest level. Blessed was recorded in Capitol’s Studio B in Los Angeles, and Was was drafted just three days before recordings began. He co-produced the album with Williams’ partner and manager, Tom Overby, and engineer Eric Liljestrand, both of whom also produced the Williams’ previous album, Little Honey (2008). The latter was Williams’ most commercially successful since her debut album Ramblin’, making one wonder why Was’ involvement was sought at all, and at the last minute. The great producer himself, however, was so keen to work with Williams he never asked her that question. “To be honest, I don’t know why they asked me!

I guess they just wanted to shake things up; you don’t want to keep making the same record over and over. I had bumped into Lucinda in January 2010 at a charity event called MusiCares, which is part of the Grammy foundation and which helps musicians in trouble, and for which Neil Young was the honouree. I was the music director and played bass, and Lucinda was also on the show. We spent some time together and got on very well. I think Tom noticed that, and made a mental note that this could be helpful later on. When they asked me to work on the album, I didn’t need any persuasion: she is one of the greatest artists on the face of the earth and to be in the studio with someone like that was an irresistible offer, especially after I’d heard the demos of her songs. She’s also an incredibly expressive singer. I’ve worked with many great singers over the years, and she’s right up there with them. You make great records with great singers and great songs.” Blessed has been released in a deluxe 2-CD version: one CD being the regular album, and the other CD containing demos of the same 12 songs, sung by Williams at her kitchen table, recorded with a Samson Zoom Q3. The album contains some of her hardest-rocking songs to date, with Williams’ striking voice in the middle, framed by guitarists Val McCallum and Greg Leisz in the left and right channel, respectively. From the demos it’s not immediately evident why any of the songs ended up being given particular treatments. When Was is asked for clarification, he seems reticent to give too many details, as if concerned about taking too much credit for the making of Blessed.

“I can tell you my perspective on how these tracks got the shape they have on the album, but that doesn’t mean I was the governing force. Lucinda had gotten together with the band before the recordings, but they hadn’t done any rehearsals and the arrangements were not yet done. I sat in the room with the musicians, put headphones on so I was hearing the same thing they were, and I listened to the vocal. As a producer I’m always listening to the vocal. When the vocal shines through, you know you’re doing the right thing. I’d heard the demos, so I knew she sounded good singing these songs alone. If you’re not enhancing that, you’re doing something wrong. You make the singer come forward, and then everything falls into place.” BLESSED RECORDINGS

Another person who can shed light on the goings-on during the making of Blessed is, of course, engineer/producer Eric Liljestrand. Hailing from New York, his credits include Laurie Anderson, Elvis Costello, John Cale and Bill Frisell, with whom he won a Grammy for Best Contemporary Jazz Album in 2004. Liljestrand is also a composer and has scored the music for several TV-shows. He recalls, “It really worked for Lucinda that nobody had played the songs until the last minute, because she doesn’t like to overwork things or sing the songs too many times. She wants to be really fresh and for the band to be really fresh also. Because Don came in at the very last minute, and we never really sat down together to discuss how we were going to do things, I don’t think anyone was sure of AT 29

what their role was. But Don usually sat with the musicians, and he has a very positive vibe in the studio. He’s like the ultimate cheerleader: he makes everybody feel good just by his presence. “We had had a couple of meetings before Don came on board, and each of us had a list of songs with annotations like: “up tempo, ballad, hard,” making reference to the general feel of the songs in vague terms. The actual shapes of the songs revealed themselves while we were working in the studio, and in some cases this took a while. For instance, for a long time we didn’t know where the song Copenhagen was going, and when we did finally capture it, we were none the wiser about it until the next day when we listened back to the tracking. We recorded all the takes – and some songs took 15 takes, others only six or seven but we nearly always used the second-last take. Don often likes that take, so much so that it became a kind of joke, as in: ‘let’s do one more, so the one we just did will be the one!’ We recorded everything on ProTools, at 24-bit/96kHz, but didn’t do a lot of micro-editing. If we did edit something, we were generally taking out or adding big chunks.” AN UFLAGGLINGLY GOOD ATTITUDE

Eric Liljestrand (above) has been working with Lucinda Williams for years, apparently. He knocked Don out with the sounds for her new album right from the get-go – in part, thanks to Capitol’s magnificent Neve 8078 console, which he describes as “a good vintage desk that’s in fantastic shape.”

‘HOME’ STUDIO MADNESS: SEX, DRUGS & (SOME) ROCK ‘N’ ROLL Don Was hails from Detroit, Michigan, where he was born in 1952 and given the name Donald Fagenson. Once he became involved in the music industry he adapted his other name, in part because he became tired of being mistaken for Steely Dan’s Donald Fagen. As a bassist, Was formed the band Was (Not Was) with David Weiss (aka David Was), which enjoyed substantial international success during the early ’80s. It disbanded in 1992, freeing Don Was up to fully dedicate himself to his production efforts. The band briefly reconvened in 2008 for an acclaimed fifth album, Boo!, and a tour. The 16 years from 1992 to 2008 saw Was’ production career reach even greater heights, during which he ran a state-of-theart studio in his Los Angeles house, called Chomsky’s Ranch. He eventually ditched this in favour of working in big studios. One of the reasons was, as he comments elsewhere, that he prefers to separate the places where he works and lives. The other was that the drugs ‘n’ sex started overwhelming the rock ‘n’ roll… Was ended up so under the influence that much of his time at Chomsky’s Ranch has now since turned into a haze, but he does recall a few things. “It was in the house next door to where I lived in Los Angeles and it was a crazy place, man. It used to belong to Russ Meyer, the director of these softcore films. It was a classic, over-the-top ’50s Californian house. It had a swimming pool that was designed so he could swim straight into the living room! I bought a Neve console and built quite a good studio, and recorded a lot of stuff there. But I was gradually losing my mind. At one stage I had lots of girls working for me and I was sitting by the pool. One day someone said to me, “you’re like Matt Helm, the character played by Dean Martin.” It just got nuts. “I did a record with Richie Sambora there and instead of focusing on music, we tried to buy these artificial suns that they use when shooting movies for simulating

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daylight. Another time we found out that a company that made trousers we liked was going out of business, so we thought, ‘Man, what are we going to wear the rest of our lives?’ so we tried to buy the company! We had this famous film director living next door, and because Richie is the loudest guitar player I have ever heard in my life, this guy would call up to complain about the noise. Since we were not zoned to be a studio, we could be closed down. So we put a big bag of reefer in his mailbox. After that, every time he ran out of pot he would complain about the noise. I thought to myself, ‘Now I’m basically a drug dealer, just to mollify this guy.’ I was a mess. I had to get back in the real world. So I sold the studio in the late ’90s, moved elsewhere, stopped getting high and made a point of going to sleep every night.” In recent years Was has also ditched the steady flow of big name artists he tends to work with. Instead, he has in recent years mainly produced relative unknowns, such as Jill Sobule, Maia Sharp, Delbert McClinton, Pieta Brown, and Elizabeth Cook. Was’ work on Blessed is his first high-profile production job in a few years. One wonders whether the man has gone all ‘idealistic’, or is taking it easy having grown tired of high profile acts and with that, perhaps, high stress jobs. The reality, however, turns out to be a striking sign of the times. Was explains: “Um… I don’t quite know how to say this, but today’s reality is that nothing sells anymore. There are no recording budgets and the royalties are insignificant. So it’s difficult to be swayed to do something that isn’t cut out for you [laughs uproariously!]. Nothing is going to sell two million copies so I just do the stuff I really want to do. I do whatever genuinely interests me, and in many ways this has made making records fun again.” So there you have it: even the world’s very top producers are finding it hard to get big paying jobs.

Don Was’ cheerleading vibe is also acknowledged in the liner notes on the album, where he is credited with “unflagging and unforgettable good attitude.” Continuing his portrayal of the sessions, Was explained, “My job as a producer is to create an inspiration-friendly environment, and one way of doing that is by casting. Casting the right people is important. I recall an actor complaining that Woody Allen didn’t give him many directions, and Allen’s thing is: “I cast the right actors so I don’t have to tell them how to deliver each line.” It’s the same with producing records. The right musicians are the ones that will understand where the artist is coming from, and they’ll have great ideas, so I don’t have to do too much. I want to be the dumbest guy in the room! When you’re working with a band it can be relatively easy, because you just want the band members to be themselves, but it can be much harder achieving that working with session musicians. But with Lucinda’s album it was easy. Everything was easy with this record. It was a f**king pleasure to make this record!” “We cut two songs a day, recorded live in the studio with Lucinda singing and playing acoustic guitar, two electric guitars, keyboards (usually B3), bass and drums. The band was set up in a circle in the main room with baffles between them, and some amps placed upstairs. They were all wearing headphones. Lucinda was in an iso-booth with her guitar, and a lot of credit must go to Eric – it’s not easy to record vocals and acoustic guitar at the same time and get a good balance. He really is the architect of the sound of that record: he did a magnificent job in a very challenging situation. I should point out that 100% of Lucinda’s vocals came from the live tracking. We didn’t overdub a single

vocal on this record. So the take was the one on which everything worked well vocally. Lucinda’s performance didn’t change much from take one to, say, take seven; it was more a matter of how we could best complement her singing. “On Soldier’s Song, for example, the puzzle was how to keep the backing as emotional as her vocal. The solution was to play less, paring down the instrumentation and putting space in, while keeping the dynamics. The extended guitar solos in the ending of Seeing Black weren’t planned. Everyone was digging the song so much they just kept going. It’s a very musician-like thing to do [laughs]. But as a producer you worry about that, because if everybody gets all hyped up, and it’s not your final take, on the next take the musicians tend to start there. Consequently, that take will be too hyped up and manic. So you don’t really want those things to go on for too long. You don’t want to burn people out either. But there was no controlling them! There were extended endings to many songs, they just happened, and people were having fun and it felt great. We did some mixes of Seeing Black that were shorter – with some edits and fades – but we immediately missed the feel of those extended solos. Finally we said: ‘f**k it, put the whole thing on!’ CAPITOL: U67s FOR DAYS

Regarding the gear with which Blessed was recorded, Was stated that he “pretty much left the sounds up to Eric [Liljestrand]. If I hadn’t liked it, I would have said something, but Eric has worked with Lucinda and these musicians for a long time, and I was knocked out with the sound from the first song.” So Liljestrand detailed the ins and outs of the gear he used: Eric: “I should preface everything by saying that Capitol has one of the best stocks of vintage mics I have ever encountered. They were bought new, have never left the building, and have been well maintained. I recorded Lucinda with a Neumann U48, which was in a box labelled ‘Frank,’ as in Frank Sinatra! I took a picture of the box because I simply couldn’t believe it. There’s such a sense of history in that building. Lucinda’s (and Frank’s) 48 went through the Neve 8078 console, which is a good vintage desk that’s in fantastic shape. All mics apart from the ones on the piano went through that board. I used some desk EQ on Lucinda’s voice, and then put it through an LA-2A before going into ProTools. There was quite a lot of spill between Lucinda’s vocal and her acoustic guitar, so I recorded the latter with a DI as well as an AKG 451 or a Sennheiser small condenser, the idea being that the DI would feature while she was singing, and the 451 would take over when she wasn’t. “Regarding the drums, I had an Electro-Voice RE20 on the inside kick and a Neumann U47 FET on the outside, a Shure SM57 on the top of the snare and an AKG C-451 or 452 underneath. The toms luxuriated in C12As, the overheads U67s, while the hi-hats were recorded through

a Shure SM81. I also had an RCA 44 out in front of the kit and a ride cymbal mic, which, from memory was another small-diaphragm AKG 451. The bass guitar was recorded with a Manley DI and another U47 FET on the cabinet. I put the mic and the DI of the upright bass through a Fairchild compressor. The upright was also recorded with a pickup and a 451 – I like the sound of a small diaphragm on a brightsounding bass. Val’s electric guitar cabinet was recorded with an SM57 close and a U67 about four feet back. I did the same thing with Greg’s amp, but in his case I used the Sennheiser 409 close to the amp. Capitol has so many U67s… I don’t like them normally, because they sound a bit brash and brittle to my ears, but Capitol’s are so beautiful-sounding that I was happy to use them. “The pedal steel went through Greg’s amp and was recorded with the same mics. The National steel was recorded with a Sennheiser and a Schoeps CMC6 with MK2 capsule, so it’s an omni. I really favour omni mics on acoustic guitars, you get a more neutral and open sound, and there are a couple of acoustic guitar overdubs done by the guys, and they all had the same Schoeps mic setups. The keyboards were a Casiotone which went direct; a Hammond organ recorded with two Sennheiser 421s on the highs and an RE20 on the lows; and the piano had (yet another) pair of 67s on it, which I moved about from song to song. Sometimes I had them positioned as an X/Y pair outside the piano, sometimes as a spaced pair, sometimes I had them rock ’n’ roll style above the hammers, depending on the sound we were after. The piano mics were the only ones that didn’t go through the Neve; for them I used Amek 9098 preamps. There was also a bit of accordion, which was recorded with the Schoeps mics. I had a booth set up with the Schoeps in it, so anybody with an acoustic instrument could jump in and be ready to go. The ‘Nail File guitar’ that I’m credited with on the album was me bowing a guitar with an emery board, played really fast with a lot of echo. I recorded it on two tracks in ProTools, and had extra long delays on each track, feeding back into each other, like a Robert Fripp technique. Finally, Capitol also has six Les Paul-designed echo chambers, two of which we used. I always tracked the chamber in case we mixed somewhere else.”

100% of Lucinda’s vocals came from the live tracking. We didn’t overdub a single vocal on this record


Blessed is one of the best-sounding albums to have been released in recent years, and has been acknowledged as such by several reviewers and various fan internet blogs. The quality of the players and singer, as well as Eric Liljestrand’s engineering skills, are undoubtedly the primary ingredients, but in addition Don Was arranged for the album to be mixed by the legendary Bob Clearmountain, best known for his work with Roxy Music, Springsteen, The Stones, Bryan Adams, Crowded House and countless others. The result is, perhaps unsurprisingly, a very rocky and muscular sound, even on the many more

Approach with Reverence: Ol’ Blue Eyes’ Neumann U48, still with his name on the box! Incredible stuff.

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Time out!: Eric and Don contemplate the next step while seated at the Neve console, during the Blessed sessions.

DON WAS ON THE PRO TOOLS ‘ABYSS’ DW: “It’s easy to fall into what I call ‘the ProTools and plug-in abyss’. I’ve certainly done that – you get hypnotised by the screen. You can descend into madness and ruin a piece of music (and your health), especially if you’re working at home and the clock isn’t ticking away your money. The law of diminishing returns definitely applies after eight hours in a studio. I don’t care what anybody says, your ears get fatigued. After eight hours you should call it quits. What happens in the next four hours won’t be as worthwhile as what happened in the first eight. And not only is it inefficient, you run the risk of destroying your songs. After eight hours you’re in a danger zone. You may get attached to things that are not the right thing to do, and working 48 hours on a song without getting any sleep f**ks you up, and it’s really easy to fall into that trap when you have your own studio. It’s better to separate the place where you make music from where you go home and sleep. You need to be able to leave and come back. “I’m fine these days working with ProTools. It still doesn’t sound as good as analogue, and digital can often sound the same, because everyone is using the same plug-ins and the same samples. But I don’t think it matters. I’m more concerned about the lyrics and the music of the songs than about that last two percent of quality that you get from tape compression. And these days, people listen to MP3 anyway. I don’t have a problem with that either really, but it certainly eliminates whatever edge analogue tape might have given you. Recording analogue is really something for the audiophiles. The other thing about ProTools is that it’s easier, and most of all, cheaper to use. When I worked with the Stones on Bridges To Babylon in 1997, we spent $200,000 on two-inch tapes! We had so many reels we had to hire a daytime and a night-time librarian, whose sole gig was to find the tapes we wanted! We ran tape all the time. Just that jam with Keith for the song How Can I Stop cost $2000! We also ran things off in stereo on that album during recording, and ended up with 75 DAT tapes. So let met put ProTools in perspective: for the Stones’ A Bigger Bang [2005], which was entirely recorded in ProTools, we spent about $1400 on hard drives! That’s it. And with today’s drives it would have been a fraction of that cost.” AT 32

meditative tracks, but it suits the music perfectly. Apparently Liljestrand, who mixed Little Honey, had also already mixed Blessed when Was suggested to call in Clearmountain. What prompted the production heavyweight to finally, umm… throw his weight about? “Eric is a great mixer,” explained Was, “so it’s nothing to do with his ability. It had to do with objectivity of every one of us on the team. I recall asking for some more room sound on one track, but it actually made it blurrier. Sometimes you want the objectivity of a fresh person coming in, and someone like Clearmountain has earned the respect of everybody before he even mixes a thing. So everybody was very hands-off while they let him do his thing, and he was able to cut straight to the essence of each song without attachment to the rough mixes that we had. I also think you want to have contrasts in ideas as well as in sound: sometimes you use Fender; sometimes you use Gibson; sometimes you record on a Neve, and also mixing on a Neve doesn’t give you as much vibrancy as when you’re recording on a Neve and mixing on an SSL.” Another thing that contributed to the excellent sound of Blessed is that the album wasn’t flattened to death at the mastering stage. It was mastered by Ted Jensen at Sterling Sound in New York, and Was commented: “I’ve used Ted a lot to master albums I’ve worked on. I trust him, and don’t feel I need to go to the mastering sessions. In any case, I don’t see the practical value. You don’t know the room, so you don’t know what you’re listening to. Also, you can

only control so much. In 1993 I was involved in the reissue of The Rolling Stones’ back catalogue for Virgin Records where we really put things under a microscope – Bob Ludwig mastered it, and it sounded great. But when we started A/B’ing manufacturing CD-lathes from different plants – even two lathes from the same plant – they sounded different. It depended on who was on the night shift, and which room they were using, and so on. Or there’d be a guy somewhere who thinks that it needs more treble. Total quality control is unenforceable, so you have to choose your battles. If you get the songs you want on the record, without the record company demanding you do a Motown cover for a single or something, you got off lightly. “I’d like to tell you that things were better in the days of vinyl, and that we had more quality control back then, but I don’t know. Historically, there are some horrible-sounding records that are among the best records ever made, because the performances and the songs are so good. Nobody compared pressings from Minneapolis with pressings from London. There were so many imperfections that you just accepted these things. Now in the digital age we expect everything to be perfect all the way down the line. But it’s not. Ultimately you have to fall back on the fact that everything lives or dies with great vocals and great songs. If it also sounds good, you got the best of both worlds.” Seems like Williams, Overby, Liljestrand, Was, and Clearmountain achieved all this with Blessed.

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MARK MOFFATT EX-PAT IN NASHVILLE Sometimes it feels like there are more Australian engineers working overseas with international acts than locals working with Australian ones. Is the grass really that much greener in places like Nashville? We find out from long-time ex-pat, Mark Moffatt. Text: Peter Moses

Mark Moffat in the legendary Blackbird ‘C Room’. AT 34

It’s a sunny afternoon in the remote leafy U.S. town they call Nashville: one of the few places left on earth where the recording industry still thrives. New commercial studios are opening every year here and the number of session musicians is well beyond counting. Long perceived as the centre of the Country Music universe – almost to the exclusion of all other musical genres – these days Nashville also boasts pop/rock powerhouses Kings of Leon, The Black Keys, Kei$ha and The White Stripes to name but a few who now reside and record in Nashville. Home and single studio facilities are abundant, bookended by the large recording complexes of Ocean Way, Sound Kitchen, Sound Emporium and Blackbird Studio, the last of these boasting the incredible control room/acoustic space of the George Massenburg-designed ‘Studio C’. Blackbird has become a haven for producers such as Brendan O’Brien, Dan Huff, and one notable Australian who’s called Nashville home since 1996 – Mark Moffatt. Right now, I’m walking down Music Row, Nashville’s three concentrated blocks of music facilities, offices and studios with the experienced Aussie ex-pat. Mark is one of the most respected producers to come out of the Australian music industry and is now one of the leaders in the emerging Nashville indie scene. With a slight Southern lilt now evident in his accent, Mark is happy to discuss his career, his knowledge and love of the history of Australian sound technology, and, of course, his creative works, including how he became one of the most trusted confidants of (and to this day personal Christmas card recipient from) Roland founder Taro Kakehashi. LONDON CALLING

PM: Mark, can you tell AT readers what first got you started in the recording industry? MM: I moved from Brisbane to London in 1972. All I’d done since school was play guitar, so I wrote to every guitar store in London and eventually landed a job at Top Gear in Denmark Street. Not only was that store considered ‘guitar central’, the street was also home to many major publishers and studios. I got to know the engineers at Central Studio and Regent Sound and would regularly duck out to play sessions there. After a while I started asking the usual ‘what does this button do?’ type questions, and during the course of this awakening Terry Britten (ex Twilights and a major writer/producer) took me under his wing and convinced me there was a career in it if I was interested. I returned to Brisbane in late ’75, and an old friend, Bruce Window – who had opened a 16-track studio with his self-built custom console and an Ampex MM1100 – asked me if I’d like to base myself there. I knew enough at that stage to mic things up and record and his technical staff graciously filled in the gaps for me. PM: What were some of your early works that are still influencing people today would you say? MM: The Saints’ I’m Stranded keeps coming up as an important record, one that’s credited with changing things worldwide in the ’70s. Over the years, I’m finding the records that pass the test of time are the ones that were ‘firsts’ rather than huge sellers. I was an early electronica experimenter and the first in the world to use the Roland TR 808 and MC4 digital sequencer on a record. At the other end of the spectrum, Anne Kirkpatrick’s Out Of The Blue album is recognised as the first ‘new country’ record in Australia. Records like these and, of course Treaty by Yothu Yindi,

opened doors that a lot of Australian artists have walked through since. That’s something I’m still proud of. STRANDED IN BRISBANE

PM: You say you still get asked about I’m Stranded. Can you tell us more about what you remember about those early guitar sounds? MM: Top Gear had all the major UK guitarists coming through every day, either to buy stuff or get repairs done. In the first few months I had tasks as diverse as scraping violin rosin off Jimmy Page’s #1 Les Paul to setting up every Beatle guitar you’ve ever seen. Watching guys like Page or Kossoff play up close, asking questions… just interacting with people like that was an opportunity of a lifetime. I learned so much about guitars, amps and sounds back then. I bought one particular amp there – a ’60 Fender Super with a Partridge output trannie that made it very loud and angry sounding – and when I returned to Brisbane, I took it with me. When The Saints came in to record, Ed had a 60-watt Vase – a great locally made valve amp – but it was too clean, so he plugged into my Super and that was that; it roared. The combo of two old Jensen 10-inch speakers just about to give out, the Partridge trannie along with Ed’s powerful right hand was pretty much the sound on I’m Stranded. There was a live track that we doubled and I also used a long cement hallway leading to the studio for distant miking. You can hear that panned against the close mic on the solo in No Time. There was also a pair of the fabled Pye compressors in the rack, which I used extensively during the session. It’s funny, despite all the rock journalist hoopla about that record, no-one in Australia has ever asked if I documented the session. Steven van Zandt here in The States asked about it recently, and as it happens, I did write a fair amount of stuff down about mics and set ups etc., which I guess will see the light of day some time soon. PM: What made you move to Melbourne soon afterwards? MM: I was in a Brisbane band, the Carol Lloyd Band (signed to EMI) and was doing all our demos at Window where I’m Stranded was recorded. The band was folding and, coincidentally, we dealt with the same A&R people at EMI who’d signed The Saints. They knew I played pedal steel guitar and asked if I would produce an ‘outlaw’ country band they’d signed called Saltbush. EMI booked TCS in Melbourne for the album, and as it happened, Barry Coburn was running TCS at the time and was really into that edgier kind of country. He kept sticking his head in during the sessions and before we finished up, he offered me a job as house engineer. So, I moved from Brisbane to Melbourne almost immediately. MELBOURNE THEN SYDNEY

PM: What was the scene like down there at the time? MM: Armstrong’s and TCS were the main facilities in Melbourne back then. Flagstaff and Richmond Recorders started a little later, along with a few other 24-track places. Mushroom Records was really doing well and Michael Gudinski kept those studios pretty busy. TCS consisted of two big old sound stages behind GTV 9 that were set up to do the music tracks for TV shows, but eventually it started doing outside sessions. There was an Auditronics 501 & MCI 24-track in Studio A, and an Optro console (which rocked!) and Ampex MM1000 16-track in Studio B. The Optro was originally in Studio A and was used for AT 35

Thinking back, I’m Stranded had caused a big stir in the industry, but I had no idea at the time that I should be using it for selfpromotion!”

Vintage gear anyone?: This collection of outboard gear is probably the most expensive per square foot ever gathered in the history of AT. Fairchilds, EMIs and Neves by the kilo.

CREDITS & CRED Not only is Mark Moffatt one of the most experienced and respected producers to emerge from Australia, he is also an accomplished musician (guitar, pedal steel), engineer, and writer. In 1980, he took a position as director of artist development/house producer with Rupert Murdoch’s Festival Records, at that time Australia’s largest independent record label with 25% of the market. That year saw Mark’s first nomination as ‘Best Producer’ and the beginning of steady chart success through the ’80s and ’90s. At the 1996 ARIA awards, seven of the 15 awards went to artists produced by Mark! During his career in Australia he featured in reader and industry polls, including the Rolling Stone ‘Best Producer’ and ARIA ‘Producer of the Year’ categories and has more productions in the APRA Top 30 songs of all time than any other single producer. As a writer he’s had two No.1 Australian country singles, two Top 10 pop singles, two Top 10 UK and European singles, and has composed scores for seven major motion pictures and TV series. Production credits include: Keith Urban/The Ranch, The Saints, Yothu Yindi, the Divinyls, Neil & Tim Finn, Ross Wilson, Slim Dusty, Tony Joe White, Stacey Earle, O’Shea, Jasmine Rae, Deana Carter, Leslie Mills (with John Shanks & the Dust Brothers), Gloriana, Adam Gregory, and Jason Aldean remixes.

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all those legendary Chain, Skyhooks & Daddy Cool records before the studios upgraded. I wish I knew where it ended up. [Answer: these days it’s owned by Melbourne sound engineer, Simon Grounds – see box item for more on the console’s recent history.] Studio A’s outboard gear included a Universal Audio LA-4, two Kepex gates, an Eventide delay, and two or three shared EMT plates; pretty sparse by today’s standards. John French was the main engineer when I arrived, and he became another major mentor of mine. In general, it was a really friendly yet competitive scene, though I was still pretty green about the business back then. Thinking back, I’m Stranded had caused a big stir in the industry, but I had no idea at the time that I should be using it for selfpromotion! (Laughs)

back then no-one liked them!). So, I went out to Channel 0 at Nunawading before I left for Sydney to receive a crash course on their Neve, which was pretty similar to the Festival desk from memory. Once I got inside the Festival business machine I began to learn more about how the record industry worked and things started moving pretty fast for me, including getting a call from Ross Wilson to produce what later became the Chemistry album for Mondo Rock, my first platinum album. But of course, in those days, once a house producer became that successful it was deemed time to leave the fold, which I hated because Festival was such a great family. I miss it still, and it’s crazy to think that the last management team eventually sent it to the wall. It had been such a well-run company before then.

PM: You moved about a bit… what made you move again, this time up to Sydney?

Some artists I worked with in those first few years at Festival include Richard Clapton, Renée Geyer, Mondo Rock, Tim Finn, Eurogliders, Ultravox and I had a huge hit with my own technopop project, The Monitors.

MM: Well, TCS management was then handed over to 3AK – the Packer owned ‘beautiful music’ station – and we all knew how that was going to pan out. Here was the funkiest, most in demand rock ’n’ roll studio in the country suddenly being managed by suits from a muzak station. I quickly got into trouble there for recording my own stuff after hours and was fired at about the same time as a friend told me about an opening as in-house producer at Festival in Sydney. I flew up for an interview and got the job. The TCS stint probably carried more weight than I realised in hindsight – I discovered later there was some tough competition for that gig. I’d never run a Neve before – funnily enough,


PM: How did your association with Roland come about? MM: I had been using their early synths in Brisbane and, as a result, got to know John Egan, the CEO of Roland Australia, who somehow saw the value in my technopop musings before anyone else and always made sure I had their latest gear. John was showing Roland founder Taro Kakehashi around Sydney one fateful day and called me at 301 saying he wanted to show Taro the studios. Coincidentally, I was in the mix

room – Studio M – that day, and happened to have one of the first AMS RMX 16 reverbs with me, which by the way, didn’t reach The States ’til much later. Mr Kakehashi mentioned they were making a digital reverb, so I asked, “Oh, will it do this?” as I dialled up 9.9 non lin on some drums. Apparently, he went straight back to the hotel that night and called the factory. I was up in Japan the next week working on the SRV 2000. That was the beginning of the relationship. I still use my SRV every day by the way! PM: What did they expect of you when you got there? MM: There were only a handful of Western people invited to participate in high-level product meetings in Hamamatsu, among them, Adrian Scott, Eric Persing (who later founded Spectrasonics), and myself. Basically, you had young Japanese engineers developing new products and they needed ‘Western ears’ and creative minds to bring professional needs and applications to the table. I was lucky enough to work closely with Mr Kikumoto, who essentially wrote the MIDI specs with some help from Tom Oberheim. Hamamatsu is an amazing place. I remember having lunch with Kikumoto one day and him saying, “Ssshh, talk softly… see that guy over there, he developed the DX7. And that guy in the corner there, he’s from Korg,” and so on. It was an eye opener to see just how concentrated the MI business was in that town, and still is. Years later, I discovered I’d been designated a ‘yatoi’ along with Tom Oberheim, Roger Linn, and other US guys. Yatoi is a 19th century term meaning: “foreign worker brought in for expertise.” SIGNING THE TREATY

PM: Treaty by Yothu Yindi was one of the nation’s most important records for more reasons than just the politics behind it. Can you tell us how that record came about? MM: The background to this story was that

I’d previously made a great record, River, with Shane Howard, and obviously, Shane was held in very high regard by the Aboriginal community across Australia thanks to his own work with Goanna. After hearing that record, Mandawuy Yunupingu had Mushroom contact me and set up a meeting. It was a long preamble because he was insistent I understood their spiritual beliefs and culture before we did any work in the studio. We eventually went into the studio in Melbourne, and I had Ricky Fataar come down and play drums given there was no drummer in the band back then. That was a big part of the whole experience for them because Ricky is such an amazingly sensitive musician. As soon as they hit the traditional vocal and didge breakdown groove in Treaty you could tell magic was happening. I loved the Razor Gang mix, of course, but the fact that the remix was the radio hit made it easy to forget the work and human effort that went into getting that band collectively on to tape in any shape, let alone something that was able to be remixed! PM: What were some of the cultural differences you had to adapt to once you got the band into the studio? MM: Shyness, a different perception of time, all those predictable things. Even then it was clear that Gurrumul Yunupingu was the big talent. He played a lot of the keyboards and guitars and sang like an angel. We became very close and spent weeks together just doing things he wanted to try, things that were in his head. I played one particular guitar lick live on the basic track of My Kind of Life, some simple nondescript little thing, but it hit a nerve with him and every time he heard it play back while we were overdubbing, he would laugh out loud and nudge me. So joyous was he. THE VAULT

PM: You set up one of Sydney’s best known studios, The Vault, at a time when private use

24-track facilities were rare. What was the motivation behind that? MM: I had always been a ‘sit at the console and play’ person and really needed a studio after leaving Festival. I bought an Auditronics 501 from AAV and imported a Soundcraft two-inch 24-track in ’82, which eventually went into a studio I built on our farm in Queensland. But ultimately the travel was too hard, so I moved the gear back to Sydney in ’86 and rented the space in Balmain. It was in pretty rough shape back then but had already been a small studio in what was once the explosive store for the Adelaide Steamship Company. It was a foot thick concrete bunker basically, hence the name. Most of the time we’d do slave reels at 301, retire to The Vault for overdubs and vocals etc, then head back to 301 and lay it all up on the Mitsubishi 32-track digital recorder to mix. It was a heap of fun and a lot of great music came out of that place. I rented it out when I finally moved to The States and sold it to James Cadsky a year later. THE BIG MOVE

PM: Why did you decide it was time to move to The States? MM: I’d hit a creative ceiling in Australia, and also I knew that in a youth oriented industry like Australia’s, there was never going to be acceptance of an Australian equivalent of Phil Ramone, Glyn Johns or T Bone Burnett, even though I was not yet that old. Not to mention that so many questionable US and UK ‘producers’ were arriving on vacation visas, convincing local A&R people they had major credits, and taking all the major album projects. More often than not, these went off the rails of course, and it was very frustrating to witness. I finally made up my mind to leave the day I saw the Neve from 301 Studio-M being crane lifted out through a hole in the wall and down onto Castlereagh Street by Fletcher’s guys [from Mercenary Audio]. I virtually lived in that room

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Many Australian artists and managers are stuck with the ‘yee haa!’ stereotype of Nashville, and fail to understand just how dense the US market is

and it was not going to be there any more. [That console now comprises half the 80-channel Neve at Ocean Way, Nashville.] I’d also like to point out here that in The States the Grammys are run by NARAS, the recording industry body, not RIAA, the record label body. Just before leaving, I was briefly involved in trying to set up an Australian producer/engineer guild, but ARIA wasn’t really interested in any dialogue with us creative types in the studio community. Hopefully, that’s changed. SETTLING ON NASHVILLE

PM: So why Nashville? What drew you to that city, not L.A. or New York? MM: Barry Coburn had moved to Nashville after leaving TCS where he had broken a huge country artist named Alan Jackson. Barry had been at me for years to move over here, and finally it seemed right. I visited in ’95 and was knocked off my feet by the studios, players… everything. Barry and I did a publishing deal and I finally moved here in August ’96. Besides any career motivation, I really just wanted to live in the music grid that runs through Nashville, Jackson, Memphis, the Mississippi Delta and Muscle Shoals. It’s tough when you’ve lived your life totally under the influence of this stuff, and one day you wake up and realise no-one really gives a shit about it in your own culture. PM: Can you talk about how you got involved with Keith Urban’s career?

SIMON GROUNDS ON HIS OPTRO CONSOLE: SG: I got wind about the Optro console back in 1990, and finally tracked it down at Melbourne’s iconic venue, Festival Hall, where it was languishing in exile in a back room like a forgotten king. It was a 16:8 console back then, with four auxes, three-band EQ and various submix matrixes – a beautifully made board. It had a nice minimalist signal path and great ’70s transformers. I’ve heard there were five made for Channel 9’s capital city studios but I can’t be certain of this, so there are others out there somewhere. I spent a long time re-capping, switching, potting and modifying the routing layout with the help of audio whiz Ross Giles, and eventually turned the board into a 20:2:2 mix console with six auxes and sub-in points on all the buses for an extended setup. I also updated the main op-amps to FETs to ‘speed up’ the sonics a bit, which has worked out well. A lot of fine Aussie artists have passed through this board since I’ve owned it, and of course there’s the rich history of bands that used it previously, as Mark Moffatt has already pointed out. Recent additions to its history include the Underground Lovers, Snout, Rocket Science, Birds Blobs, Laura Jean, Guttersnipes, KES, Spoils, Black Pony Express, Shower Scene from Psycho, Sandro, Anyones, Dave Graney, and Panel of Judges to name but a few. The Optro lives on!

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MM: Barry Coburn and his wife Jewel had done very well, establishing a thriving publishing and management business, and had signed Keith to a publishing contract. When I arrived, I really didn’t have a clear picture of what my role would entail, but as it turned out I was mainly employed to work with their writers and artists, Keith obviously being among them. This pretty quickly developed into an A&R role inside what had already become the most creative and supportive environment I’d ever experienced. It wasn’t exactly easy for anybody back then, and there was a lot of wheel spinning on Keith’s part, but he was given the space to get through his stuff, and once he did, things started to fall into place very quickly. There are any number of people in Nashville willing to take credit for Keith’s rise, but the family at Ten Ten Music had an enormous amount to do with it. PM: How would you describe Nashville as a music and studio city to people who have never been here? MM: Well, there’s simply nothing like it on the planet, and the last couple of years has seen it emerge as the industry hub across many genres, to the point where even Rolling Stone magazine now claims Nashville has the best music scene in the US. Besides these very positive changes in the local business, the big attractions here are the location and lifestyle – 65% of major US markets are easy drives away, something that’s impossible from LA or NY. Nashville apartment and housing rents average US$700 – $1000 per month, vans and trailers are cheap, and there’s a concentrated

yet accessible creative community like nowhere else, especially for musicians and writers. The quantity and quality of studios speaks for themselves – where else can you find 35 Elam 251s and 16 channels of Fairchild under one roof? When I did the Digidesign Strike drum sessions in Studio D, Blackbird owner John McBride pulled out all the stops with the outboard gear and wheeled in pieces I’d never even seen before, like the EMI tube curve bender. These days that kind of thing can only happen here, and maybe Abbey Road. Having said all that, breaking into the producer/ engineer side of things here can be a long haul, and it was like starting from scratch for me when I first arrived. It was very tough. PM: What are you working on currently? MM: Currently, I have a band with Derek St Holmes who sang all those great Ted Nugent hits. We’re recording right now with a couple of guests, including Brad Whitford from Aerosmith. It’s a total DIY project, so we’re tracking at home and it’s sounding great. I mixed some Gloriana tracks for Matt Serletic last year and they’ve called about mixing some more just this week. PM: Nashville-based Aussies O’Shea recently had a No.1 country song here; can you tell us more about them? MM: I produced Mark O’Shea’s first record for ABC Country, which did well, but he was very young and pretty quickly veered off down a more rock oriented path. We lost touch for a while, but then when he moved from LA to Nashville, I hooked him up with some major writers who liked him a lot, and he was accepted quickly into a normally very closed group. His wife, Jay, moved here a little later, and they started doing a weekly gig as individual artists. This quickly got huge, and they morphed into a duo not long after. They entered a CMT reality show, Can You Duet, and came in fourth, which gave them massive US TV exposure. So, we went in with their live band and recorded last year, all self-funded. Their circle of incredibly supportive co-writers includes Shawn Colvin, Kim Carnes, Billy Falcon (Bon Jovi), Georgia Middleman and Dave Berg (Keith Urban). Needless to say, the songs are fantastic. PM: Anything you’d like to say before we wrap this up Mark? MM: I see so many Australians dismiss Nashville, focussing instead on the ‘meetings in LA/NY’ mentality, and going down the SxSW route, and then wondering why they’re not getting anywhere in the US. Many Australian artists and managers are stuck with the ‘yee haa!’ stereotype of Nashville, and fail to understand just how dense the US market is and that short visits here don’t really achieve a whole lot. At some point you need to be on the ground here. Besides being a unique place to write and record, this town has become a very affordable and hip base – a stepping off point for touring artists. The Australian industry, in general, would do well to look in to that aspect of the town.


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In this two-part exploration of stereo panning techniques, Andy Stewart resuscitates the mixing tutorial concept last spotted in Issue 69, and explores how panning helps recreate three-dimensional space. Text: Andy Stewart

It’s a big wide world – the world of stereo panning – both literally and metaphorically. Surround sound is immeasurably more complex, of course, but for now let’s discuss what takes place between two speakers only. Before I begin I must stress that whatever is explored in this article represents only a small fraction of the ideas you can apply to panning. Personal exploration and experimentation are the true keys to the art of mixing, and the best teachers are you and the speakers in front of you. Think laterally, listen, imagine, and trust your instincts for what sounds right. Don’t forget, you’ve grown up in physical space all your life, so you know more about panning than you perhaps realise.


In this first installment of a two-part investigation, we’ll be looking at panning in the physical sense: how it helps create the illusion of space and time, and how it affects the scale of a mix. Next issue we’ll explore other panning fundamentals and delve into the creation of more radical effects-oriented techniques. For now though, let’s look left and right… horizon bound.


To me, panning is one of the most powerful tools in a mix engineer’s arsenal. It may seem obvious to say this, but before stereo, there was no such thing as positioning a sound in the space between two speakers. In fact, even after stereo entered the picture there was still no such thing, only a switch that sent a channel’s signal to left, centre or right.

So what is stereo panning all about and how does it figure in a stereo mix? Well, I should pause here for a moment and ask you to go outside and ponder this question yourself. The answers are literally everywhere, and analysing the world around you and how you react to it, is a great way

‘SEEING’ IS BELIEVING Not always, but often, I like to mix with my eyes as much as my ears. I’d like to think there were countless situations where this concept doesn’t really apply, but the more I mix, the more I find myself imagining the space in front of me. To me, building a mix is like painting a large picture, though not always. There are things in the background, characters in the foreground, lesser information at the corners of the canvas, things presented in stark focus, others less so and so on. Next time you’re in the great outdoors, note that information at the extreme edges of your vision tend to be out of focus, and louder sounds naturally encourage you to turn and face them. AT 40

to learn how placement works in combination with tone, reverberation, and dynamic range. PHYSICAL REALITY

Fundamentally, stereo panning is about placing a sound in a context, and that context is, in this case, our two speakers. What’s really happening to an audio signal when it’s panned is that more, equal amounts, or less of its voltage is being sent to the left or right speaker, creating the illusion that sound is coming from the far left, the far right, the middle, and so on. A guitar, for example, panned left-of-centre in a stereo image is simply coming out of the left speaker at a greater volume than the right; pan it centre and it’s coming from both speakers equally. Particularly in this second instance, if you had to point to where the instrument was in the stereo image, you’d point to the space in between the two speakers where no speaker exists – the ‘phantom centre’. To that extent, all panning is an illusion. With this in mind, let’s explore some physical spaces and look at how panning helps propagate this illusion.



If mixing, for you, is fundamentally about creating a balanced three-dimensional world for your audio signals to inhabit, imagining that space from the outset is very important. Your early vision for it may evolve into something else later, of course, or be overrun by a stronger idea that occurs to you midstream, but imagining how the space you’ve chosen might behave if it were real is a good place to start.

Let’s say the task at hand is mixing a conventional rock song. A good thing to consider early on is how large the space will ultimately be that the overall mix is trying to occupy. Typically, the more sounds there are, the bigger that space will need to be – though not always. The way a song is recorded might also determine how you pan the signals, but whether it’s a tight, dry and airless room you’re looking to generate, or an epic Blue Mountains landscape, the sooner you decide this the better. Either way, choose your physical illusion wisely, in sympathy with the story the music is telling, but remember, if you have 100 people standing in front of you – and you want to see them all – your approach to panning is going to be quite different to if there had only been five or six (more on this shortly).

Okay, so let’s say we’re going for the ‘Blue Mountains’ option. I’ve heard the vocal, listened to the lead break and am inspired to go for broke! There aren’t 150 sounds that need accommodating between the speakers but nevertheless I want an epic soundscape. The decision is made, and it feels good… CREATING A FOCUS

Time for some quick decisions… though these don’t necessarily need be set in stone quite yet. For the moment, the key is to remain open: to suggestion, to accidents, to ideas – any spark that will help trigger the illusion of space. First up, let’s decide on our fundamental focus. I’m going to say it’s the main vocal at this point (surprise, surprise) and pan this mono source dead centre right from the get-go, along with the bass guitar and kick drum, as per convention, and so that my bottom-end is being played by two speakers pushing and pulling in unison. Panning the focus elements dead centre also appeals to the listener on an unconscious level because almost anyone who focuses on a sound in the physical world will simultaneously turn (or attempt to turn) towards it, so the sound source is symmetrically positioned directly in front of them. (There’s no time to go into detail here about the physiology of this phenomenon, but simply stated, the desire of the brain to focus on a sound triggers the body to turn towards it – so that eyes and ears are front and centre.) Other instruments in my mix are two electric guitars and an acoustic, along with a tambourine, a keyboard drone, two BVs of the fairer sex, and several other incidental percussion and string-based instruments. I’m not sure what to do with the guitars just yet but my early instinct is to pan the two electrics to about 8.30am and 3.30pm on the pan pots (about 87% on the DAW panners – more on why I’m not pushing them out to 100% in a moment). I’ll keep the acoustic guitar in the middle for now, it being the odd one out and one of three guitars. Incidentally, all these stringed instruments have been recorded with two mics, and they’ve not been submixed, which means I have the option of panning them in different ways, or not. As things progress I settle on the panning of the electrics, and pan the stereo acoustic mics to about 10am and 2pm (about 40% on the DAW). One of these has slightly more

bottom-end than the other, making the image seem slightly lopsided in favour of the brighter mic, and being obsessed with stereo balance – which we all should be – I endeavour to close the tonal gap between them a bit with EQ tweaks before finally deciding to pull the brighter mic in a tad. Lots of other things are addressed during the course of the day, including the lead guitar break: the close-amp mic channel of which is panned centre with its room mics panned 60% left and right, and its level comparable to that of the vocal, befitting its focus – the other electrics are still playing during the solo. STEREO ELECTRICS

As the mix evolves my mind is constantly addressing the picture in front of me, and the epic nature of the landscape I’m trying to portray. The guitars and vocal are key to this landscape making sense so I hone in further on the stereo electrics. These two instruments are interacting well and are naturally creating some nice movement across the stereo image, but the left one is my favourite; it’s big and beautifully played, and has epic qualities written all over it. The two mics on this instrument are an AEA R92 and a Shure SM57 (let’s say), and to create the illusion of scale, I change the position of the 57 to about 3pm on the pan pot and turn its level down a fair bit. I also add to this a delay of about 100ms (mixed 100% ‘wet’ with about 30% feedback) and immediately the scale starts changing. I exaggerate the 57’s tonal response, making it slightly boxier and harder, turn it down even further and immediately the guitar has started to sound ‘a bit epic’ – thanks to the panning, the altered tone (a sound at distance never has the same fidelity as when it’s up close) and the delay. But then it dawns on me: my big landscape guitar now sounds too much like it’s inside, not outside... the slap echo is too distinct and wall-like, even though it can certainly sound like that in the great outdoors sometimes. Hmmm. I mumble something to myself like: ‘yeah, but who cares, stop being a super-realist you idiot’ but change tack anyway. I swap the mics around: the 57 is now 87% left and the R92 ribbon takes over the 57’s more distant, right-of-centre role, including taking on its delay settings, which are themselves now sounding duller and rounder. Suddenly my guitar is clearer and harder sounding; too hard in fact, so I ditch PHANTOM CENTRE


(Line of symmetry)







Simple & Clear: Simpler mixes allow the elements within to be bigger and wider. Less instruments means more space: space for improved fidelity, more width for instruments, and increased transparency. In the example above, the vocals can be big and




full, the electric guitar can occupy most of the left side and the piano most of the right. Acoustics are mid-panned and everything is free to overlap and sink deep into the background courtesy of panned reverb and delays.

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(Line of symmetry)







VOCAL & BVs More Sounds = Less Space: In our slightly more complex ‘Blue Mountains’ mix, the elements within are panned for balance and width, but each instrument and voice is now occupying slightly



less real estate. Symmetry remains critical. Things that overlap are placed at varying depths, and reverb and delay help create the illusion of space. Tambourines and synth pads are placed

the 57’s initial EQ setting. I add a big 480L plate reverb, fed by the ribbon mic – panned hard left/ right and featuring nothing much above 900Hz. The ribbon mic’s rounder tone accentuates the dull yet lengthy nature of the 480L space. To this I dial in a 180ms predelay. I eventually decide to duplicate the ribbon mic channel to help broaden and deepen the 57’s tone by tucking this duplicate back in behind the 57, separated in width by about 10%. Bingo, the guitar has become more vivid but also more epic sounding – close yet enormous, thanks to the panning and predelays creating the illusion of something big and vaguely reflective in the middle distance on the right-hand side, and behind that a dull, wide and deep backdrop. There is no ‘zing’ coming back to the listener from this particular outdoor landscape. The acoustic gets a smidge of the guitar’s 480L, but stays fairly tight in with the main vocal, which has a complex array of compressors, automated EQ and a vocal reverb. The vocal also has two delays on it that are panned 70% left/ right – very dull – one of which is made to sound like it’s bouncing off the same hilltop as the

THE MIND’S EYE If you’re sitting down at your DAW or console (or both) and preparing yourself for a mix session, in most cases, you’ll be starting with a digital multitrack file either of your own making or someone else’s. The files will nearly always be a combination of stereo sounds, discrete mono instruments, groups of instruments etc, all in a relatively shambolic state. When you first pull up new session files, you’ll often curiously find that they’re quite bunched up in the middle, like a giant fur ball. This often indicates several things: that there are lots of mono sources panned centre (pretty obviously because they’re not mixed yet), that the stereo pairs feature the source in the centre, and that panning is going to play an important role in the mix. Even if there are lots of stereo pairs in the multitrack, these will still most likely sound mono in amongst all this clutter because they will nearly all have been recorded by engineers who have falsely assumed that stereo pairs should always be recorded symmetrically. In reality, it’s often preferable that the stereo sound be recorded asymmetrically to emphasise the space around it or give it character.

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in the background and project forward, while guitars and the main vocal occupy the front grid positions, echoing and decaying into a big, deep and dark background.

guitar by also receiving a smidge of the 480L, and suddenly the song comes to life visually. I pan the drum kit but limit its width to a maximum of about 55% left/right on the overheads and toms. It sounds better from the kit’s individual perspective to go wider, but it makes no sense visually for it to be this wide in the physical landscape. When it’s made to sit back eight yards or so, thanks to a reverb, yet is panned hard left/ right, it’s suddenly trying very hard to be 80 feet wide! I don’t want that. By placing the drums inside the guitars, their proportion remains realistic and the environment is made to sound even bigger, not smaller. Sorry drums. USE YOUR ILLUSION

Finally (and briefly) to our other sounds: the female BVs are made brighter than the main vocal – yes brighter – but smaller, wetter and wider – panned to about 8am and 4pm, along with their reverb. I later just go ‘bugger it’ and pan their reverb fully left and right. The tambourine is back in the mix, panned centre, and its plug-in insert reverb is about 90% wet with no predelay and panned tight, about 35% left/right. This makes it seem like it’s far away. The key to this illusion is understanding that a big predelay would have inferred that the tambourine was closer to us than the mountains behind it – a predelay tends to separate a sound from the environment around it – but we don’t want that. We want it to come towards us from a great distance as if it’s out there in the landscape. No predelay helps convince us of this ‘natural phenomenon’ and contributes to the overall illusion of scale. I won’t go into any more detail than that – hopefully you get the picture. The key to this mix being both interesting and physically ‘outdoorsy’ sounding is by constantly asking yourself questions like: ‘How would the ‘Blue Mountains’ themselves have reacted to this combination of sounds, where is Signal X originating from, and how should that affect my panning, tone and reverb?’ and so on. I’m not

suggesting you become a literalist about all this, nor am I inferring that this disciplined visual approach applies to every mix – not at all. I’m simply illustrating the point that understanding how an actual landscape might have responded to the band had they literally been there, helps to provide your mind’s eye with a framework in which to work. MONO E MONO

Above all else, panning helps create space and balance. The real trick is determining how to pan all your different sounds and instruments in such a way that the overall outcome remains symmetrical. One way to do this relates directly to the concept of panning for scale. Let’s say, for example, we’re mixing 250 channels of instruments, voices and sound effects – I know, I know, that’s a hell of a lot! At the beginning of such a daunting session, it might seem nigh on impossible to ascertain what goes where, particularly when the multitrack file is delivered to you like a ‘fur ball’, with most things bunched up in the middle (see The Mind’s Eye box item for more on this). Trying to find ‘like pairs’ and ‘opposites’ in this world of confusion is exceedingly difficult. In this extreme case the most important first principle to grasp is that the mix is going to be big – and unless you cut 100 channels out, you’re in for a very long haul. Secondly, a mix involving this many ‘characters’ means that, in some form or other, most of them will need to be discrete – i.e. mono (or almost mono). Yes, I know there are precious, masterfully captured stereo recordings of pianos, synths, choirs, guitars, orchestras, and so on in amongst all this, but trust me, if you get hung up on each one’s exquisite individuality – as it was recorded – and try and find a big wide space for each of them to occupy, you’re going to get very frustrated very quickly… and nowhere fast. All you’ll end up with is a big fat mess right in the middle of the image. The main alternative to this is to edit 60 percent of the instruments out of the mix – a viable, and in some cases

preferable, alternative. (If an artist insists on having 250 sounds in their mix, it’s vitally important to make them realise early on that each sound will compromise all the others, and when there are so many, the compromises will be significant.) Generally stated, big mixes featuring very few instruments can be easily created using wide panning regimes and stereo placement of instruments, delays and reverberation. Big mixes featuring hundreds of instruments, meanwhile, mostly require pinpoint mono sources, as well as layers of depth, a fine grasp of width, and overall balance. Discrete sources create natural focus, a vital ingredient when so many sounds are vying for the crowded space. If you’re faced with this level of mix complexity, consider the ‘mono sources’ option. Somewhat counter-intuitively, 200-odd mono sources panned throughout a stereo image will typically sound truly enormous, provided they are also placed at various distances from the listener. Lots of stereo instruments panned left/right, on the other hand, will not. BALANCE, BALANCE, BALANCE

Panning is also fundamentally about balance. Bright signals panned left require some sort of counterbalance in the right. Things soloing in the right speaker should be overtaken by another instrument soloing in the left, and so on. It’s bad panning practice to make a mix sound lopsided for extended periods of time, either by making the tonal balance seem skewed to the left or right

– too bright on one side – or by making one side inadvertently louder or more dynamic than the other. Whatever your instruments, and regardless of how many there are, it’s vitally important to achieve balance with your panning. If the music draws you in one direction for some reason or another, the balance must naturally be rectified by the next musical movement that comes along. Sure there are exceptions to this dogma, but they are few and far between. A mix that makes someone feel like they have something wrong with one ear, or speaker, is a bad mix. It might seem ‘cool’ at first, but that impression won’t last. YOU ARE THE PERSPECTIVE

I won’t delve into another example now – we’re out of room unfortunately. Instead, I want to leave you with one final notion that applies more to physical three-dimensional panning regimes than the more radical ones we’ll explore next issue. When you’re creating visual, illusion-based mixes like the one touched on earlier, there’s one thing to keep in mind: you, the listener, are the perspective. If you’re creating a big mix – very three-dimensional, wide and deep sounding – don’t push instruments that you want close up, too wide. The reason for this is fairly straightforward yet subjective, but also dependent on how much you want to outright mimic principles that govern the physical world. If a sound (in an otherwise big, three-

dimensional mix) is dry, yet panned 100% left, it feels like it’s stepping outside the lines of the stereo image (see illustration). Without any signal coming from the right speaker, it no longer feels like it’s in the space, but rather separated from it. If that’s what you want to achieve with this particular sound, fine – just be aware of it. The drier the sound, the closer it theoretically seems to be. If it’s 100% left, not only will it tend to compromise the scale of our most distant objects, it will also start sounding like its coming from behind you, particularly in headphones (more on that next issue). Not all mix engineers would agree with this, but in general, a dry, full-fidelity sound panned hard will tend to fight against the illusion of our deep, three-dimensionality, since no reverb can exist outside it. It also makes a listener instinctively want to turn towards it, which is impossible, and in headphones, would set you spinning! Avoid going past about 90% if you insist on going ‘wide and dry’. Of course, if you simply want to confound the overall illusion, go right ahead. Only one rule applies in the end: do you like the sound of it? Next issue we’ll get stuck into some of the ways panning can contribute to movement and depth, how an instrument’s pan position might be anticipated at the recording stage, and explore a few of the more ‘out there’ effectsbased techniques. Until then, explore the space (particularly with a cowbell), and always remember, balance is king.

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When they ‘dry out’ some can do more damage than a bushfire. So why do capacitors fail, and which ones are the most susceptible? Text: Rob Squire

Here on the workbench there’s one electronic component above all others that I deal with on a daily basis. Whether I’m refurbishing an old ’50s tube limiter, repairing console channel strips or hunting down an intermittent crackle in a microphone, capacitors are typically the star of the show, infamous for causing a broad range of faults in all sorts of audio equipment. Indeed, there’s probably no other component used that’s subject to more discussion, debate or diatribe than the capacitor, so this issue, let’s grab a handful and get our heads around what they are, and why they’re so demanding of attention. CAP IN HAND

Roll out the barrel!: This is what happens to an electrolytic capacitor when it’s installed with reverse polarity, neatly revealing the rolled-up construction.

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A capacitor – more commonly referred to as a ‘cap’ around the traps – is a device that stores electrical charge, consisting of a pair of conductors separated by an insulator. The ratio of this stored charge to the voltage between the conductors is a measure of the capacitance of the device. A set of hi-hats is another type of capacitor, albeit a very poor low-value capacitor providing very little capacity to store charge, where each cymbal is a conductor and the air between them an insulator. What’s required in a real capacitor is for the conductors to have a large sheet-like surface area, be very, very close to each other, helped along by an insulating material sandwiched between them. The insulating material, also called the ‘dielectric’ can consist of a number of materials: air, paper, various plastics, mica, ceramics and metal oxides, each of which has a quite distinct set of properties.

The dielectric constant, or ‘K value’, of each material fundamentally affects the amount of capacitance available for a given spacing and surface area of the conducting sheets. The chosen material also determines the temperature stability of the device, its behaviour around water and heat, and ultimately the general name given to the capacitor type. Thus we refer to ceramic capacitors, mica capacitors, and polystyrene capacitors, and so on. Maximising the amount of capacitance in small caps usually involves either stacking alternating layers of conductor and insulating dielectric materials on one another in a multi-storey sandwich configuration or rolling up alternating sheets of conductor and dielectric, jam-and-sponge-roll fashion. Caps exhibit rising impedance with decreasing frequency, in effect reducing current flow as the frequencies descend until ultimately at 0Hz – better known as DC – they have infinitely high impedance resulting in all DC current flow being blocked. In audio applications this fundamental property leads to caps being used for three fundamental circuit functions: blocking DC voltage from one circuit element’s output from the input of the next (a cap in this situation is called a coupling cap and is in-line with the signal path); power supply decoupling and storage, where the capacitor is placed across the power rails around circuit elements to ensure noise-free and stable operation; and frequency selective circuits such as filters and equalisers, where caps are exploited for their frequencydependent characteristics.

Above: Circuit for a high-pass filter. Left: Different capacitor types: Left to Right: Aluminium Electrolytic, Tantalum, Ceramic, Polypropylene, Polyester, Polystyrene, Mica.


The most common capacitor type is the aluminium electrolytic capacitor. These caps are constructed from two conducting aluminium foils, one of which has an insulating oxide layer, with the foils separated by a spacer soaked in conductive liquid electrolyte. This is rolled up, placed in a cylindrical casing and fitted with two connection pins, or legs. This formula is cheap to manufacture and yields a cap with large amount of capacitance for its physical size. It’s polarised, meaning that any DC voltages applied across it must be orientated in accordance with the positive or negative markings on the body of the capacitor. The only other capacitor commonly available that cares about this DC orientation is the tantalum capacitor – other plastic and ceramic types aren’t concerned about their orientation with regard to DC voltages. Aluminium electrolytic and tantalum capacitors that have a reverse DC voltage applied to them will be destroyed by the experience (see pic, left), and if there’s enough power on hand at the time, often quite explosively. The liquid electrolyte in this capacitor, while contributing to its cheap construction cost, ultimately leads to this type of capacitor’s downfall. Heat and time cause loss of the electrolyte: the cap ‘dries out’ – a familiar term to many – becoming less and less of a capacitor as time goes by. This leads to the daily grind of vintage audio repair and refurbishment: locating, testing and ultimately replacing dried out caps. TO RECAP OR NOT RECAP

I’m often asked to speculate whether the caps in unsighted ‘Unit X’ need replacing. Sometimes I get the more direct request when a unit is sent in for service: “replace the caps while you’re at it can ya please Rob?”. I’ve poked a stick at hundreds of thousands of capacitors in my time, leading to some rules about capacitors and their replacement, but like all good rules, occasionally these get broken and at other times confounded by a new experience. When an electrolytic ‘coupling cap’ in a signal path dries out two things happen to the audio passing through it: the overall level will be attenuated, and, in some instances, the low frequency response will be further depleted. These two symptoms of a degraded capacitor – loss of level and loss of low frequency response – are due to the cap developing a high resistance, and thus behaving more like a resistor, and the value of the capacitance decreasing. In any device that houses lots of caps in its signal path, such as a mixing

console, this sort of degradation leads to strange variations in level and low frequency response from channel to channel, and output to output. From an operator’s point of view, a console in this state feels less precise and develops an aura of sonic uncertainty. The thing that makes developing rules about replacing capacitors tricky is that different brands, and even different value caps within a single brand, degrade at vastly different rates. Even the physical position of a capacitor on a circuit board can affect its rate of degradation. Recently I was checking over caps on a very late model SSL 6000 console, and on one circuit card found three particular electrolytic caps of the same brand and value in each channel. In every case, across dozens of these channels, the same two capacitors tested fine while the third was badly degraded. The reason for this bizarre ‘consistent inconsistency’ only dawned on me after I positioned the channel strip vertically in its normal operating position. The degraded cap was located above half a dozen ICs that ran reasonably warm, whereas the two undegraded ones were located below these ICs. Simple convection (heat rising) had done its job on this one cap, the lower ones remaining just that much cooler and thus avoiding the most common killer of electrolytic caps – heat. This pattern was broadly spread across all the channel strips, where capacitors higher up on these large and deep circuit boards, had degraded more than those lower down. TIME CAPS

Time yields hard won experience, they say. Standing back and surveying the myriad brands and values of capacitors used over the years reveals how some brands and values within brands have stood up to the stresses of heat and time better than others. A prime example of this is the blue axial capacitor, manufactured by Philips from the ’70s to ’90s. This was a capacitor used by the truckload by UK and European manufacturers of pro audio gear. Found inside the Neve 51 and V-Series console, which run very hot, these capacitors have proven to be a cap that degrades very badly, sometimes fatally in these environments. There wouldn’t be one of these consoles left working on the planet that hasn’t had all its capacitors replaced by now. Indeed, those that made the mistake of replacing the capacitors with the same brand the first time around have now had to replace them all over again. Ouch!

ESR? A capacitor also possesses a small amount of resistance, effectively appearing in series with its capacitance. Not surprisingly this is called ‘equivalent series resistance’, or ESR. In a non-electrolytic capacitor such as polyester this ESR is very small, usually less than 0.1Ω. However, in aluminium electrolytics the ESR is much higher and is dependent on the capacitance, voltage rating and design of the capacitor. Some electrolytics are designed and sold specifically as low ESR types, and while they’re typically employed to decouple switchmode power supplies, they also prove to be very good as coupling and decoupling caps in audio circuits. The trademark feature of a capacitor that’s ‘drying out’ is an increase in its ESR. This elevated resistance in a coupling capacitor is what causes signal level loss. In assessing capacitors to determine their current condition the ESR meter is the go-to tool. Simply measuring across a capacitor will quickly indicate its ESR leading to conclusions about if and how far the component has degraded. In most instances this test can be performed while the cap stays in its unpowered circuit, making for rapid testing. While commercial ESR meters are available, for 90 bucks and a couple of hours assembly time, you can build your own with a kit available from Aztronics. I have two and they’re used daily!

For capacitors housed in these hot environments there are now, mercifully, electrolytic capacitors that have a higher AT 45

temperature rating, meaning that their expected lifetime at a given temperature is increased. These 105º caps, as they’re called, are not much more expensive that the more common 85º caps, and generally speaking there are very few good reasons not to use them. I will always use 105º caps wherever possible in any device that runs at a temperature much above ambient. Of course I can already hear yelling from the back row… “What about the sound? Some capacitors sound better than others and I’ve been recommended these special audio capacitors that are made on the forge of Zeus!” Well yes… and no, and maybe. Like many of these audiophile perspectives, there’s a touch of science and a mountain of mysticism involved here. Different types of capacitors used in different types of circuits can have an appreciable influence. Here a designer schooled in the science will be able to choose the right capacitor for the right job. Certainly capacitors such as polystyrene and polypropylene have very advantageous properties in circuits such as filters and equalisers or in high power circuits such as passive speaker crossovers. But since these types of caps don’t degrade with heat like aluminium electrolytics they aren’t the usual candidates for replacement. Also, their comparatively large physical size for the same given capacitance makes them physically unsuitable to fit into circuit boards where electrolytics were previously installed. Thus the argument often revolves around the best electrolytic brand type for the job. Enter the snake oil salesmen. My rule is simple: if there’s heat around use a 105º-specified capacitor of an established brand you can afford. If choices need to be made, a low ESR [see the box item for an explanation of this term] capacitor is generally more versatile, meaning it is suitable as either a coupling cap or decoupling cap and may mean you only have to get one type instead of two. When replacing the thousands of capacitors in a large console the fewer types you have to deal with the better. DSITORTION SPECS

Back in 2002 Cyril Bateman published a series of articles in Electronics and Wireless World magazine that was an exhaustive scientific study on distortion behaviour in capacitors. The series kicked off with the design and construction of a distortion analyser to conduct the tests, as nothing was readily available at the time that could probe the minuscule levels of distortion found inside capacitors. Ultimately he developed a system that could measure distortion down to 0.00003% and proceeded to conduct over 2000 distortion tests on various types, values and brands of capacitor. The broad conclusion of his work was that polypropylene capacitors were the best performers, and you’ll now see these type of capacitors used in highend loudspeaker crossovers and where smaller value high voltage caps are required. For example, Avalon Designs uses these types of capacitors in the signal path wherever DC blocking is required.

Some degraded electrolytic capacitors removed from a Neve 51-Series console! AT 46

For the more common aluminium electrolytic capacitor, distortion performance wasn’t as good but nothing measured much worse than 0.005%. This puts distortion factors well down into the noise floor of most equipment and is comparable with distortion factors of other circuit components. The one electrolytic capacitor that stood out as having generally better performance was the bipolar electrolytic capacitor. This capacitor is essentially two

capacitors placed back-to-back inside one case. It doesn’t care about the DC polarisation like other electrolytics. When musing over replacement capacitors this type can be worth pursuing; the drawbacks being that they’re physically larger, more expensive and harder to source. Some manufacturers also see the value of the bipolar electrolytic. For example, the SSL 6000 console mentioned earlier uses several of them, whereas earlier versions of the console used the more standard polarised electrolytic. TANTALISING TANTALUMS

Tantalums are also a commonly found capacitor in audio equipment. They became all the rage in the 1970s due largely to their physically small size for a given capacity and excellent very high frequency performance. Despite also being a polarised electrolytic capacitor they are most often manufactured with a solid electrolyte and thus don’t suffer from ‘drying out’ like aluminium electolytics, which extends their service life considerably. However, tantalums are very sensitive to the application of reverse polarity DC and current surges and when they die they don’t go gracefully, usually failing as a dead short. Worse, since they’re often used as decoupling capacitors placed across power rails this dead short invariably leads to other parts failing down the line, the chain reaction in some cases being quite catastrophic. History has shown that it’s rarely a good idea to use tantalums as a decoupling capacitor, especially with aluminium electrolytics now available exhibiting equivalent high frequency characteristics. The other quite common failure mode of tantalums is to become ‘crackly’, and frustratingly, often intermittently so. I had a U47FET in the workshop recently that would spit out a brief crackle once or twice every hour giving me uselessly narrow windows of opportunity to locate the faulty part. In the end simply replacing every tantalum capacitor in the microphone was the only expedient solution. Many condenser microphones use tantalums, taking advantage of their small size, and these parts are probably the most common source of faulty mics – whether that manifests as a ‘dead’ mic or an intermittent crackle. Tantalums are certainly one capacitor that raises debate about the contribution of capacitors to audio quality or euphonics. Various studies – including Cyril Bateman’s – demonstrate that tantalum capacitors do have significant distortion qualities, and given their tendency to fail either to a short circuit or an annoying crackle, it’s always tempting to replace them. However, one place where you’ll find these capacitors used extensively is in Neve products from the ’70s and there is little doubt that replacing all the tantalums in a Neve 1073 does alter its sonic qualities. Given this, capacitor replacement – especially changing the capacitor type – should never be undertaken without reference to the role the capacitor plays in the circuit. As a rule I tend to replace tantalums used as decoupling capacitors with low ESR aluminium electolytics to avoid the grief of a dead short tantalum down the track, but always replace tantalums in coupling circuits with fresh tantalums. For those of you heating up your soldering irons for a DIY project or to refurbish that big second-hand console you recently bought for a song, certainly give thought to the types and brands of capacitors used. Good brands of high temperature, low ESR aluminium electrolytics will be suitable in 90% of situations. Before you fork out the dosh, understand the role the capacitor plays in the circuit first and replace like with like, unless you understand the ramifications of any changes you make.

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Sound absorbing mattresses everywhere, old Akai reel-to-reels, ’80s Roland and Yamaha keyboards, bed posts and an awful lot of percussion overdubs – it must be Megastick Fanfare’s debut album! Greg Walker caught up with the Sydneysiders and their producer, Frey Lindsay, to get some insights into their densely layered sound. Text: Greg Walker Photography: Alex Coutts

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I caught up with Adam Zwi and Danny Keig from Megastick Fanfare as well as co-producer Frey Lindsay recently to talk about the recording of the band’s debut album, and it quickly became apparent that the project had been a true labour of love, taking a fair bit of time to complete. With a limited toolset of instruments they created rich layered sounds thanks to a large amount of external processing and re-amping, and the end result is an album with its own unique sound. But are they still speaking to one another? I had to find out… Greg Walker: Congratulations on the record guys. Can you talk us through the process of tracking the new album and how it evolved during the sessions? Danny Keig: We recorded most of the album at Adam Zwi’s place, which is also where we rehearse. We had a nice setup there where we deadened the downstairs room a little, and had a multicore running upstairs to Frey [Lindsay] in our makeshift control room. In the first three or four weeks we did about 70% of the recording, and when the house was occupied again we relocated to a room in the city. Unfortunately, that only lasted for a few weeks before we ended up back at Zwi’s again. After that there were a few more broken months of overdubbing, followed by

critical listening and over-analysing at the end. Frey Lindsay: Danny and I went to school together and we were always making vague plans and talking shop (very amateurishly) so we were all good friends before we started. I’m actually responsible for the name Megastick Fanfare, something I am eternally sorry for. We’d done a few demos and I liked the sound of the room, not totally dry but not too dead either, so we got stuck into it pretty quickly. We did it all on ProTools LE 8 with a Metric Halo 2882 feeding to and clocking a Digidesign 003. The wordclock on the Metric Halo is far superior to the one inside the 003, and that certainly helped. I also made sure I used the inputs on the 2882 first and only used the ones on the 003 if I needed more channels. We used a lot of digital/ electronic equipment but tried to maintain a solid analogue front-end and get the sounds going through real spaces (sending sounds to guitar amps and recording the ambience or re-amplifying/saturating with old valve amps) to give everything its own character and add a bit of grit. GW: Was it the kind of record that was all planned out in advance? FL: Actually, the whole process was very ad-hoc. We spent a lot of time messing about with sounds

and trying stupid ideas like letting the monitors feed back into the drum mics or tracking bird noises outside (don’t think they made it in). Some of it worked – a lot didn’t. There was no real grand plan for the record. We just really wanted to make something unique that occupied a different place sonically to what other people were doing. GW: How much live ensemble recording was there as opposed to tracks built up bit by bit? DK: There are a few tracks where we recorded as an ensemble, sometimes with drums, bass, guitar and keys. We were pretty lucky that we had these two little cupboards/rooms downstairs where we could isolate amps. There was an enormous amount of overdubbing though. That’s where the songs really came alive. The majority of the looping was also performed, in the sense that it’s not a triggered loop that’s structured and plays throughout a song. Tracking the loops was really just like tracking any other instrument. Adam Zwi: We also tracked the pre-recorded samples we’d previously used in our live shows straight from our samplers in the same way we would perform them live – same with the vocal loops I’d sample live at shows on my Line 6 DL4. FL: As Danny said, we did the bulk of the recording in the first few weeks. The band had been playing these songs live for a long time and really had their performances down so we were able to put down a solid foundation straight away and then build on that. As a general rule, we’d always have drums and bass playing together as a bare minimum, to get the rhythm working. We’d usually split the bass signal to the amp and an API 512 DI – we struggled a lot with the bass cab actually, mostly because it was in another pretty small side-room and we could never get the low end working right. Shaun Grevler plays a lot of higher-octave melodic bass lines and we would often just use the DI signal and re-amp it to something else to give it a bit more ambience, which seemed to work pretty well. TAPE LOOP TOPOGRAPHY

GW: There’s a lot of tasty synth work on the album. What were your weapons of choice there? DK: At the time, we didn’t have much available synth-wise. All we had was this crummy digital Roland keyboard that I bought a few years ago, yet in the end that little Roland really defined some of the songs. It really said a lot. There are some other textures on the album that are often mistaken for keyboards that aren’t actually synths – although they’re synthesised in a sense – such as my cassette loops or Adam’s vocal effects. Everything is really effects-based to give the sounds a space or a different tonal quality. We recorded a lot of this stuff direct while other elements were run through outboard or amps. GW: I’m curious about these tape loops – do tell. DK: One day I learned how to chop up cassettes and turn them into mini tape loops. I ran these

through a four-track Portastudio and use some of the features on the desk to change the sound. I also use a handheld tape deck sometimes. All the stuff on the loops is sound we record onto them and then alter with effects. The handheld can be heard on T is a P, where there are some weird vocal sounds coming out of the handheld’s inbuilt speaker and into my guitar pickup (see pic overleaf). The handheld has an awesome pause switch too that really squishes the sound as it stops. In Brain Tooth you can hear these weird pitchy tremolo guitar crescendos, which is the sound of building them up and then dropping them two octaves on the Portastudio. GW: How did you deal with all the synth action Frey? FL: There wasn’t too much on the recording side there, mostly all the synths went clean into API 512s. I didn’t mind that old Roland keyboard; all the sounds were pretty stock but they seemed to fit well. In terms of post-production, we used the Soundtoys Echoboy and Crystallizer a lot, as well as the SPL Transient Designers (analogue and plug-in). Other gear included the ubiquitous Eventide H3000, Yamaha REV-100 digital reverb (which has a really nice lo-fi graininess to its sound), Great River EQ-2NV and a huge amount of pedals. The [Line 6] DL4 was, as Adam says, really integral but we also used the Frostwave Resonator filter, Boss DM-2 delay and CE-1 Chorus, and the Moogerfooger Ring-Mod, Low-Pass Filter and MuRf a fair bit. The effects being so important, we spent a lot of time getting everything right, re-recording the tracks with the effects laid on and balanced just right so it was all locked in – no messing about later with ‘wet’ and ‘dry’ channels. By the time it got to Jono Boulet for mixing, we were sure all the important elements were sounding how we wanted them to, and he did an excellent job of bringing out the sparkle in everything.

Above: Danny Keig runs some cassette loops through the four-track Portastudio to ‘change the sound’. Below: The outboard rack had some tasty equipment in it, including a pair of UREI LA-4 compressor/limiters, an API 3124MB+ four-channel mic preamp/mixer, Metric Halo 2882 converters, and two Sytek MPX-4A four-channel preamps.

AZ: We also owe a lot to the Yamaha VSS-30 sampling keyboard. They are incredible little machines. I don’t understand why they don’t make things like this any more. You feed any sound, great or small, into the tiny hidden microphone and it comes out toasty warm and ready to be squished, stretched or manipulated in any way you like (the vocal sample in Good Phier is a good example of this). We tried tracking it straight from the output, but the tiny speaker distortion was half the charm so we ended up miking it with an SM57. VOCALS & EFFECTS EXTRAVAGANZA

GW: What was your general approach to recording the vocals? AZ: The original intention was to record vocals live but we couldn’t isolate them enough. Something we are going to have to work on for future recordings. DK: In the end we did all the vocals as overdubs. We had a mattress-based vocal booth for most songs. There were a whole bunch of microphones AT 49

Clockwise from Bottom Left: The handheld cassette recorder was used to replay and re-record vocals via its inbuilt speaker and a guitar pickup; Frey Lindsay plays some extra keyboards between production and mixing duties; Sam Goldsmith on the skins in the downstairs recording space.

THE MIXER’S PERSPECTIVE – JONATHAN BOULET SPEAKS: The boys recorded it all in ProTools, but I’m a Cubase man myself, so Danny had the fine job of bouncing out every single layer into Wav files. And good god, there were a lot of layers! That was the hardest part about it, making space for everything and just knowing what parts were to take priority. There were so many lines weaving in and out, adding little flourishes and that kind of thing. I think I learned a valuable lesson about how all the little differences add up. Just little EQ tweaks here and there would help so much further down the track. I’m really into layering lately. There’s definitely a lot in there and it does that thing where you hear new things each time you listen to the record. I used Cubase Studio 4, Waves Gold plug-ins and monitored on Yamaha HS80s and AKG K240 Headphones. I don’t have any outboard gear, mainly due to financial restrictions. So the whole thing was mixed in the box. I’ve been told that I live in plug-in land by other engineers before but they were wankers. I’m very much a supporter of the “it’s the ear not the gear” mantra. I’d rather learn to use what I’ve got really well instead of use some ritzy high-end compressor badly. It’s like giving a monkey the keys to a sports car. On a more personal level, these guys are Class-A champs and fit for reproduction. Would I work with them again? No, they’re all crazy. Stay the hell away from them!

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we used for vocals. Some memorable ones were an Audio-Technica AT4040, an old RCA 74-B ‘Junior Velocity’ ribbon, a Legacy Axis 70 (budget Royer 121-styled ribbon clone) and various Beyerdynamic mics such as the Soundstar and M69. Frey has some beautiful mics. In fact, Frey has so much great gear it’s ridiculous! FL: Not any more, I pawned it all to get my sister out on bail! DK: All of our vocal effects were performed live through re-amping, because it would have taken way too long to program all our changes/warps and it helped maintain a human feel. GW: Sounds like there was a lot of re-amping going on. Did you have any preferred gear for this task? FL: Re-amping is a fantastic technique. A reamping box is basically a reverse DI, converting line-level back to guitar-level so you can send the sound out of the computer through whatever you like: pedals, speakers, anything you can plug a jack into. The possibilities are really endless, and it’s great to be able to design your own atmospheres and effects instead of relying on the standard gear. We used a Musicman amp a lot; it has such an amazingly crisp sound, and I sent quite a few drum tracks out through the little built-in speaker on an Akai tape machine and rerecorded it for a rough kind of ambience. There

were some other little tricks but we honestly tried so many silly ideas that I can’t remember which ones stuck! DK: It wasn’t just amps either. We also used lots of guitar pedals and other crazy stuff. I distinctly remember Frey running a snare track through a reverb, into this big tube tape machine, driving it pretty hard, and then recording what came out of its speaker. The other reason why we would re-amp was so we could perform the effects over what we had recorded, to give more control over that aspect of the performance. THE ENDGAME

GW: Earlier you mentioned the dreaded critical listening and over-analysing phase at the end. Do you think good things came from it or was it more a case of trying not to lose perspective? DK: There were definitely positive things that came from it, but it’s largely why it took so long to finish up. I think we all went through a period of over-thinking things and adding too much so we just had to tone it back a little bit. We had to sit down together and pull out many of the excess elements before we sent the album off to be mixed. We only kept the things that added to the image or purpose of the songs. AZ: I think we could have benefited from a logistical deadline too. Our manager was in Mongolia at the time and there was no label or

You lose all perspective and momentum and suddenly that track you were so excited by a few weeks ago is the soundtrack to your own madness!


distributor in sight, so there wasn’t a real sense of urgency to get it done! Having said that I’m really happy with how it sounds now and I guess it would have sounded different otherwise. FL: That stage is the absolute worst. It’s like a nightmare twilight zone other-dimension where six people sit in a room for days at a time, listening to the same loop over and over and over again, and thinking they’re hearing something different each time. I think this is the most detrimental thing that can happen to a recording; you lose all perspective and momentum and suddenly that track you were so excited by a few weeks ago is the soundtrack to your own madness! It really is crucial to keep your focus on the whole mix and not the individual elements. Another engineer, Wes Chew (who mixed the original Braintooth single) taught me to avoid soloing individual tracks (barring some technical reason such as looking for a buzz), always keeping a few other elements in there for contrast and balance, otherwise you end up with an instrument that sounds amazing by itself but completely disappears in the mix. So we really tried to step back all the time or even work on a different song for a while just to keep it from stagnating.



GW: There are some great clicky sounding percussion elements that recur throughout the album. Are these generated acoustically or electronically or is it a combination? DK: I’m pretty sure all of these were acoustic in the end. We did heaps of percussive overdubs, with all of us in a room banging stuff or with Zwi sitting in front of the computer, possessed by the percussion demon. There were countless instances of arriving at Zwi’s to find yet more percussion tracks had been added. AK: I’m pretty sure everything in the room during the recording sessions was hit at some point or other, including the bed frames we dragged in. FL: I am still amazed by some of Zwi’s percussion, especially on Teething. The way he improvised and played with the various possibilities was really impressive. I’d just follow him around with a mic, usually that Legacy Ribbon; he’d tap on various surfaces – piano lids, pieces of metal, small children etc – until he found what he liked. Great fun. Pretty much every percussion sound you hear was done live by him – he’s the original analogue Beat Machine. I did some hand-claps too; I think they really tie the album together. In terms of how it all came together, I think we did alright, although I won’t be speaking to any of them ever again!

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Got any news about the happenings in your studio or venue? Email Brad at or go to the AT website and register online:

Text: Brad Watts

Kicking off with the Victorian sound-smiths this issue, we see Toyland Studio in Northcote busy recording basic tracks for the upcoming Bloodduster album, and completing an album for Melbourne death metal band, Eskhaton. At the other extreme, jazz fusion band, Logic, mixed their Live at Bennetts Lane DVD, Indy band, Wild Mind, mixed their full length album, hardcore band, The Day Everything Became Nothing, tracked to two-inch tape (Ampex 456 running at 15IPS) for their upcoming album, and AOR band, White Widow, tracked guitars for their second album. Toyland also completed production audio for Radio 3AW and the Tutankhamun exhibition. Radio and TV spots include the Led Zeplin Experience and Bat Outa Hell shows. Gear used (and abused) at Toyland includes the ever-faithful Neve 1084, Focusrite ISA 110 mic preamps and original UREI 1176 compressors. At Crystal Mastering, those ensconced on the Crystal Couch included The Peep Temple, Black Water Riff, Scarlett, Brothers Grim, Jericco, Andy Van, A Day in the Life, Marvin Priest, In Hearts Wake, Maria Ford, Bak, and The Paper Kites. Haunting August also rocked up with mixes crafted by Magoo. Subwoofer cones were flexed courtesy of Phaze One, The Funkoars, The Herd, Fluent Form, Chico Johnson, and Jools. Mikey Young was also at Crystal, mastering a compilation and some rare gems from Eddy Current Suppression Ring. Russel Clarke was in with an album engineered by Mark Ludwig. Down at the beach in Altona, Coloursound Recording Studio played host to Ballarat’s own Tessa and the Typecast, who have been busy

recording two singles with engineer, Callum Barter. Meanwhile, Mat Robins is in the early stages of tracking an album with Melbourne band, Cardboard Machines. Coloursound is also inviting artists, and freelance engineers to pop in to say hello and see if the studio can cater to their next project. T-Bone Tunes Recording Studio in St Kilda has been busy with tracks recorded and produced for March In Moscow, Barry Jones, Dee Ellus, Karl Fredericks and The Broken Splendour. New acquisitions abound: most recently (due to popular demand), mastering plug-ins Izotope Ozone and Waves Center. There’s also a new microphone stand that looks like a crane, apparently. T-Bone proprietor, Tony Salter, reckons it’s ace for overhead drum placement – it holds the Neumanns perfectly. Audrey Studios has been working on Liz Stringer’s album, with Craig Pilkington producing. Indigenous rock legends, Coloured Stone, dropped in to do a new version of their classic hit, Black Boy. Eddie Perfect tracked and mixed a song for the TV series, Offspring, Craig mixed and mastered an album for alternative country band, The Stillsons, and Alex Hallahan has been mixing his next album. Van Walker is building a compilation of his early albums, and tracked a new tune. Indie pop greats, The Crayon Fields, laid down some rhythm tracks for an album, and The Killjoys have been recording their next album involving all the members of the band from throughout its 23-year history. Audrey also pounced upon some new preamps – the AMS Neve 4081 Quad – which is doing amazing things for the vintage ribbon mics on the ‘LOZ’ setting.

At Sample and Hold Studio, the crew completed audio mixes for TV campaigns for Disney movies, Pirates of the Carribean: On Stranger Tides and Cars 2, sound design for short films The Wardrobe and White Square, along with recording and mixing songs with singer/ songwriter, Ben Usher. David Ashton is also pleased to report that a podcast he recorded and edited, Boxcutters Episode 246, has won a Chronos Award. Props to David! Now to deepest, darkest Tassie, where Digital City Studios’ Michael Gissing has been in a flurry with no less than four feature-length documentaries, back to back in the lead-up to the Sydney Film Festival. Two documentaries: A Common Purpose and Scarlet Road were completed just weeks before being premiered at the SFF. A third documentary, Trouble with St Mary’s was screened on ABC’s Compass. The final project for May/June was Curse of the Gothic, which was mixed in Melbourne at Soundwaves by Michael. The doco traces the 28-year struggle to perform Havigal Brian’s epic work, Symphony Number 1 – The Gothic, in Brisbane. With a 150-piece orchestra, four brass bands and a choir of over 300 voices, the epic two-hour symphony was finally performed at QPAC Brisbane last December. The doco features part of the performance plus details of the alleged ‘curse of The Gothic’ – intriguing. Michael’s also added a couple of sheep paddocks to his farm (baaah). In Sydney, Omegaman is kicking arse with his new 12-inch, Skankin’ Riddim plus remixes for Washington DC act, Empresarios, on Fort Knox Recordings – check Soundcloud for the lowdown.

Opened in 1998, Rancom St Studios in Botany, NSW was the result of a long search by owner/producer Garth Porter and engineer Ted Howard to create a studio space with big open rooms, easy access and ample parking and no aircraft noise! The added bonus was a stable for Garth’s unrivaled collection of vintage guitar amps and assorted vintage studio equipment. Rancom St. features a large control room with plenty of natural light. The main recording space offers incredibly high ceilings and three isolation rooms – ideal for entire album projects – from tracking through to mixing and mastering.


A unique aspect of Rancom St is the ability to combine analogue and digital. The MCI JH16 24-track can be phase-locked to ProTools. Tracks can be recorded on two-inch tape then dumped in sync to ProTools for overdubs.

During mixing, the 24-track is switched back on with ProTools chasing for the best of both old and new. ‘Straight to ’Tools’ is also a viable option with the variety of A/D convertors on offer, including conversion units from Prism, Apogee, Digidesign and RME. Garth is an unashamed gearaholic, and the gear listing reflects his fine taste in vintage equipment. A quick look around the control room reveals preamps such as the revered TAB V-series units (V72, V76, V276 – 17 of them in all), along with all the usual compression suspects – originals of course. Plus there’s ample vintage Neumann and AKG microphone action, with everything meeting at the MCI JH 600 40-channel console. Rancom St Studios: (02) 9695 1059 or

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Over at Spacejunk, the timEbandit tracked the debut album for A Winters Wish – ambient post rock with acoustics guitars through Leslie cabinets with echos into echos into echos into... you get the picture. Mr Bandit also mixed the second single and EP for Sydney electro-pop kids, The Khans, and completed a nine-day album recording block for Brisbane’s Montpelier. Tricks discovered included: recording the Mojave M200 through an Empirical Labs Fatso Jr with every button engaged and every LED active – on everything; a Chinese Bardl PZM boundary mic on the top skin of the snare drum; while room sound of the year goes to a pair of Audix OM7 dynamics placed in the corners of the room with the rear ends on-axis with the snare drum. These were fed left and right into the Fatso in ‘Everything Mode’, via the Toft channel EQ. Mr Bandit’s been using it on… well, everything. Rumor of the month: an MCI JH24 two-inch machine might soon be floating around with the other spacejunk. King Willie, a.k.a. Commander Stryker, a.k.a. Will Bowden, of King Willie Sound, has tamed albums from Gotye, Graeme Connors (mixed by Matt Fell), Ellesquire, Ru C.L, Kirsty Aikers (mixed by Paul Greene and Matt Fell)), Made in Japan (mixed by Woody Annison), D.I.G, Caitlin Park, James Valentine Quartet, Fuego Lento (mixed by Bob Scott), Lucie Thorne, Dragon, Seeker Lover Keeper (vinyl and remixes), The Lurkers, Wade Jackson, City Riots, Scarlett Affection, The Dirty 30 (various lunatics), Dom Gannon, and Murder of Crow. There were also singles and EPs cut for Charlie Mayfair, Betty Airs, Ranger Spacey, Dana Hassal, Bastian’s Happy Flight, Newman Duo, Sista Burly, and Horrorshow. The mastering maestro claims there’s others he’s forgotten! The man is a machine. Sydney’s Studios 301 has been mastering The Living End, Eskimo Joe, Wim, Art Vs Science, Short Stack, Sneaky Sound System, Powderfinger, John Butler Trio, Fourplay, Clare Bowditch, Tex Perkins’ ‘Band of Gold’, Deep Sea Arcade, Midnight Oil, Brian McFadden, Birds of Tokyo,


and Stonefield, along with remixes for Lady Gaga, Kylie, Darren Hayes, Faker, Owl Eyes, and Havana Brown. Recording and mixing was completed for Taio Cruz, TV Rock, Mr Barry Morgan, Taikoz, Cameras, Delta Riggs, Catcall & The Jezabels, Brian McFadden, as well as Tim Carr mixing The Herd and Jonathan Boulet, with Michael Morgan mixing Cameras and U.K. act, Kids in Glass Houses. Blair Joscelyn of Nylon Studios confessed to sneaking out of bed at 4.30 a.m. to secure the winning bid on a 1978 Korg VC-10 vocoder – a gorgeous and frightfully rare specimen that will co-habit snugly alongside his Korg MS-20. Speaking of snug co-habitation, Blair tells us he was ambushed by his wide-awake significant other upon his return to the sack, enquiring as to his excitement and why he’d been on the computer. Blair shrugged off the inquisition with a nonchalant, “oh just buying a keyboard on eBay.” Mrs Joscelyn wasn’t too pleased to hear his winning bid was in the $900 range – he’s not confessed to her quite yet that this figure was in Euros! Damien Gerards has welcomed Noiseworks into the DG den, John Kennedy has started his ‘Sun Studios Sessions’ covers project, and Lucy De Soto of Pete Wells (RIP) Band fame will be bringing her band in to track with Sir Russell Pilling. Recent session highlights include The Grand Lethals recording another single, and northern beaches electronica/alternative band, Kinetic Luke, completing their debut EP with Russell. Moving way up the coast to Queensland, where the locals are beginning to whine about the supposedly inclement weather, it’s been a busy couple of months for DOMC Mastering. The past two months has seen Sam Grace from the Optimen with a five-track EP, the DieVsCity crew with their latest hip hop EP, and Heliport Studios has started to bring in some amazing tracks from James, the Heliport engineer. Regurgitator have returned to the facility for final touches on their next album, and Metal Wrath have come in with tracks from their next

screamfest. DOMC has also been doing swags of tape-to-digital transfers. Thanks aplenty goes to Izotope RX2 Advanced. DOMC has also been auditioning a prototype from JLM Audio: an analogue MS decoder featuring some hefty transformers. Song Doctor Productions recently held a grand opening weekend at its D4Studio – a big success apparently, with live performances, beers and a barbeque. Queenslanders and their inclement weather, eh! Proprietor, Matt Dever has started a project with an organisation called Mununjali House, providing support for aboriginal youth in the district. Song Doctor Productions is recording traditional aboriginal chants complete with didge, clapsticks, and all sorts of other traditional percussion. Matt’s also been on the road doing FOH for Beyond The Darkside, the Pink Floyd tribute show touring with 80,000 Watts of 5.1 surround. In between shows, Matt’s assisted with transforming an old house into a community radio station, complete with a powerful little recording studio. The gang moved the local community radio station, Beau FM, into the building without skipping any airtime. Claiming he has spare time, Matt has started two Pultec kits from Drip Electronics – hit him up for an appraisal via the AT forum! Finally, flitting over to Adelaide, Sascha Budimski has had a busy few months. Sascha has created scores for a variety of theatre works including Three Men in a Tub by Jason Cross (which has just returned from a tour of South Korea), MineEnemyMine by Brisbane based choreographer Zaimon Vilmanis, Involuntary by Katrina Lazaroff at the Adelaide Festival Centre, and updated and remixed his original score for Alice Holland’s work, Phoenix, for Steps Youth Dance Company in Perth. Sascha managed to fit in a week of lecturing at University of South Australia’s Magill campus, teaching students how to get funky and experimental with audio software and a MIDI keyboard. Recent acquisitions include a BeesNeez Arabella GT mic, a Grace m101 preamp and Native Instruments’ Komplete 7.

Pony Music is a busy place. The studio has actually acquired more equipment than it has space for, so a new outboard rack system has recently been installed. Some of the items going into the ‘enhanced rack’ include a second ELI Distressor, a Vintech X81 mic pre/EQ, an Orban Optimod, Lexicon PCM70, Yamaha SPX90, Rev5 and SRV2000, and a soon to be refurbished Byer 77 to add to the Byer 66 open-reel tape recorder. Pony also recently installed a Neve 542 summing console with 12 stereo channels. Mics-wise there’s an array of Neumann ‘U’ series delicacies, along with some tasty contemporary mics from the likes of Mojave Audio and AEA.

50 electric guitars, 40-odd acoustic guitars, 20 basses, 30 instrument amplifiers and over 100 guitar pedals – more than enough toys to create music in the 10 x 8 x 5.5m recording space.

Recording at Pony Music also gives you access to all instruments the facility keeps on hand. In addition to the studio smorgasbord, there’s over 12 drum kits,

Pony Music: 0412 354 254 or

Proprietor, Damien Young has recently finished tracking an album for Mr. Holy Grail himself, Mark Seymour, with some great musicians including: Peter ‘Maz’ Maslen (Boom Crash Opera) on drums, Cameron McKenzie (Horsehead) on guitar and John Favaro (The Badloves) on bass. Damien is also often on the road doing live shows and recording Leo Sayer – primarily using Avid/Digidesign consoles and multitracking every show he can.


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PC AUDIO If your audio performance droops at low latency settings, is it your PC or audio interface that’s to blame? Text: Martin Walker

If you wanted to run your PC with really low audio latency, until a couple of years ago you either had to budget for a faster CPU or accept significantly reduced performance compared with higher latencies. In benchmark tests, most of the processor families from both Intel and AMD would work beautifully with buffer sizes down to about 256 samples, which equates to around 6ms playback latency with a sample rate of 44.1kHz, making it passable for many keyboard players playing softsynths, but offering a barely workable 12ms round trip buffer latency if you needed to monitor incoming audio with plug-in effects.

sadly the Core i7 series occasionally exposed a weakness elsewhere in the chain. Some of the audio PCs I tested still ran unexpectedly poorly with buffer sizes of 128 samples and under, but since I already knew how capable the Core i7 series was I tried to track down the cause, which turned out to be various models of supplied Firewire audio interface. Once I had replaced them with my 10-year old standby PCI-based audio interface I, on occasion, managed to run twice as many simultaneous plug-ins with 64-sample buffers, as well as gaining the option of using 32-sample buffers, which these supplied interfaces lacked.

However, with buffer sizes lower than 256 samples, the CPU overheads of a 2008-vintage PC would start to rise and the maximum number of plug-ins/ softsynth voices it could run would start to drop. My tests showed that performance might drop by perhaps 10% with 128-sample buffers (round trip 6ms), but typically by 30 to 40% with 64-sample buffers, and by 60% or more with 32-sample buffers. Many musicians were keen to work with 64-sample buffers for a round trip latency of around 3ms, which makes it realistically possible for vocalists to listen to themselves in headphones while performing (the holy grail of latency), for guitarists to work with plug-in effects in ‘real time’, and for electronic drummers to experience the joy of ‘tight timing’.

Various industry insiders suspect the Firewire controller chips inside the audio interface to be the guilty party, since most audio interface manufacturers only get the choice of two or three chip models from a couple of OEM companies (apart from RME, who tend to use their own design of Field Programmable Gate Array [FPGA] chips, and who regularly top the results for driver efficiency at all buffer sizes). It doesn’t help performance comparisons when some interface manufacturers add ‘hidden’ playback buffers on some of their models to ensure smooth audio delivery, but which add to overall playback latency, while others simply double the size of playback buffers beyond your choice without telling you.


This scenario changed when Intel introduced their Core i7/i5 series. I was amazed when I tested my first audio PC with the popular Core i7 920 CPU, since its performance only drooped by a tiny 3% at 128 samples, a modest 8% at 64 samples, and under 20% with a tiny 32-sample buffer size. Suddenly you didn’t have to sacrifice much performance to experience the thrill of being able to hear yourself perform in ‘real time’ with a host of yummy software effects, or have to choose the sometimes fiddly alternative of using an audio interface offering ‘zero latency’ monitoring to bypass your computer delays and use temporary hardware or DSP effects instead. All should have been sweetness and light, but

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So, beyond my personal tests, plus anecdotal grumblings on various forums worldwide, do we have any other hard evidence for this ‘drooping Firewire performance at low latencies’ phenomenon? Well, actually we do, since my old friend Vin Curigliano (of DAWbench fame) has recently been exploring this area for himself with a test PC and a batch of different Firewire interfaces from several manufacturers (along with a PCIe reference card), producing some very sobering results (visit http://forum.dawbench. com/showthread.php?1548-Audio-Interface-LowLatency-Performance-Data-Base for more details). His measured differences tie in closely with those I’ve experienced over the years, proving that if you want to run your PC at low latencies your choice of audio interface can sometimes have a huge effect on the results.

Neither Vin nor I are damning any particular audio interface manufacturer here; performance can vary from model to model in any range, and the choice of third-party Firewire controller chips seems very limited, sometimes with older chips providing better low-latency performance than more recent alternatives. Moreover, newer Firewire models from one manufacturer at least (M-Audio) have demonstrated significant improvements in low-latency performance, so it’s not all doom and gloom. THE FINAL QUESTION

Choosing an audio interface has never been an easy task, since whether you’re a newbie or a seasoned pro there’s now a bewildering choice of audio interfaces available at prices ranging from pocket money to arranged loan. The most important question remains ‘What do you want to record?’, since this determines how many simultaneous inputs, and of which types, you’ll need, while the number of outputs will determine whether you can run a 5.1 surround setup or plumb in external hardware effects where there’s no software equivalent. Extras such as digital and MIDI I/O, headphone outputs and even DSP effect and modelling options may also be important to your way of working. Overall, the ‘best’ audio interface for a musician creating electronica may also be totally different from one used to record a live band, while if you only require stereo inputs and outputs, within a given budget you’ll generally get better audio quality than spending the same amount of money on an interface with eight or more channels. However, if you intend to go the Firewire route, you may now have another consideration. Keyboard players can often safely ignore it, as can many sound designers and those using their PCs mainly for mix-down and mastering, but for those who need to hear what’s coming into their computer with scarcely perceptible time delays, low-latency performance may now become another crucial factor in your final choice of audio interface. Isn’t life complicated sometimes?

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MAC AUDIO Now is not the time to splash out on a new MacPro. Text: Brad Watts

ith the processing brawn available from Apple’s Intel i7 and i5 MacBooks, MacBook Pros, iMacs and Mac minis, it’s seldom I speak to anybody considering the purchase of a MacPro – the Big Mac of Apple’s hardware world – the most ‘professional’ incarnation of Apple desktop computers. With the proliferation of Firewire, USB2 and more recently, USB3 audio interfaces and Firewire-based DSP-assisted units, an iMac or MacBook will tackle most audio work you care to throw at it, and this is usually where most people will invest their computing budget – they’re also far less expensive. Generally, the only reason to shell out gobs of cash for a MacPro monster is the need to install PCI cards such as UAD2s and ProTools. Other factors include quieter graphics processing and simply the desire for the fastest machine possible. With a six-core processor running at 3.33GHz, or eight cores at 2.4GHz, through to the behemoth 12-core MacPro running at 2.93GHz, a MacPro will blow any iMac into the weeds. Perhaps another consideration for audio professionals is the MacPro Xeon processor’s ability to address far more RAM than the iMac’s i7 processor: 64GB as opposed to a mere 8GB – definitely a consideration when migrating toward 64-bit DAWs and plug-ins. So there you go: a few issues to ponder if you’re tossing up the difference and can stretch the budget to accommodate a MacPro – it’s a $700 leap from the top-of-the-range iMac to the lowliest MacPro. RUMOUR MILL

The reason I mention these differences is partly due to the impending release of newly designed MacPro machines. The rumour-mill suggests we’ll see newly coined MacPros around the end of July or early August – it could be worth waiting a few weeks before you take the plunge as rumours also suggest there will be a number of changes in the new models. So what could actually supersede a 12-core Mac Pro? I’m glad I’ve posed the question for you. The Xeon Westmere processor is well over 18 months old now, so it’s likely we can look forward to the new generation of i7 Sandy Bridge processors being lumped into the MacPro. It’s worked for the MacBook Pros, so we can probably speculate these will be the processors found in the new MacPros. These will of course be faster, beginning at around 3.1GHz. Sandy Bridge is, however, a departure from previous processing architectures so let’s hope there aren’t too many teething issues with audio related software and processing. I’ll wager we’ll see Thunderbolt connectivity also. A 10Gb (gigabits) per-second transfer rate is precisely what professional desktop machines should be brandishing. Thunderbolt is 12 times faster than Firewire 800, 20 times AT 58

faster than USB 2.0, and much faster than USB 3.0. Plus we’re seeing audio manufacturers pick up the standard almost as fast as Thunderbolt was announced: Apogee, MOTU, and Universal Audio are all hitching a ride on Thunderbolt’s wagon. There’s also speculation regarding a revamp of the MacPro enclosure – that Apple is narrowing the width from its current 20cm to approximately 13cm – enough to allow the Mac Pro to be housed in 19-inch racks and take up ony three rack spaces – this could sound the death knell for MacPro server machines. TIGHTEN UP

In the meantime, I wouldn’t be rushing out to buy any of the current MacPro models. Stick with what you’ve got for now. Until then, here’s a few things you can do to squeeze a little more juice out of your current machine, and this applies to any Mac running OSX. Install Applejack and run it over your system. I know I always mention this but it’s the most foolproof way of deleting swap-files, checking disc integrity and repairing your system drive permissions. Google it – it’s free. Next, make sure your system drive has plenty of space. OSX uses hard drive space as virtual memory when there’s not enough actual memory available. As a rule, make sure there’s at least 10% free. To gain a bit more space you can remove any languages from the system you don’t need (or understand). Monolingual is a free app that will do the job quite well. Macaroni will do it also. Remove any apps you no longer use and keep your Dock free of apps you seldom use. While you’re doing this it’s a good idea to clean up your desktop – either put the stuff that’s hanging around on the desktop into folders or bung it in the trash – a cluttered desktop will slow down OSX. Disable things like Internet Sharing, Dashboard and Exposé, and any animation effects that may be turned on in the Dock preferences. All these procedures are free and shouldn’t take you more than an hour to complete. If you fancy shelling out some coin for some extra efficiency, put as much RAM as you can afford into your machine, and for some exceptional operating system speed, grab a solid-state hard drive (SSD) and reinstall or clone your system on the SSD using Apple’s Time Machine or Carbon Copy Cloner. You can pick up 3.5-inch 120GB SSDs for as little as $200 nowadays, and 256GB models for around the $400 mark. Sure there won’t be enough room for your iTunes library but there should be enough for your main applications such as DAW applications and plug-ins. Give it a whirl – you’ll be amazed by the speed increase.

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TRITON AUDIO D2O FET MIC PREAMP 75dB of total gain (although it strangely reads 50dB and 70dB). There’s also a button to flip between two transformer ratios of the Lundahl 1538 input transformer – 1:2.5 or 1:5. Besides altering voltage gain, this also switches the input impedance between 2.25kΩ and 9kΩ. Out back are XLR in and outs, and a balanced 1/4-inch jack output, as well as an input for the 24V AC power supply. It’s a tidy package, and you’d not be miffed in the slightest by the overall build quality – remarkable in this era of pressed metal, get-it-out-the-door audio gear. The device is also available in a 500-series form-factor. I’ve long been a fan of Siemens and Telefunken V76 preamps. Manufactured in the ’60s by the Germans, who were task masters of fine tolerances and top shelf broadcast equipment back then, as they are now, the V76 is guaranteed to give you 76dB of quiet, pristine gain, and make any mic sound like a backdrop to the pearly gates the second you arc it up. Unfortunately, they’re a rare commodity these days, and will set you back a fair wad of cash – and that’s before you have them repaired and racked. So it’s perhaps no surprise – in this day and age where audio equipment of yesteryear is forever being reverse engineered – that

a manufacturer has picked up this established vintage design and evolved its own take on it. Triton Audio has dubbed its version the D20 Fet Mic Preamp.


touch and once fully powered could possibly discolour your best teak or mahogany sideboard! The DACport garners all its power entirely from a USB port, ramping up over a few minutes to eventually be powering two 9V rails.

And it’s done a nice job. My first impressions of the D2O FET mic preamp are that it’s a solid, well-made piece of microphone amplification. The casing is made of astoundingly solid 4mm thick cast aluminium, finished in a utilitarian grey similar to that of its European ancestor. Like the Siemens V76, the D20 offers 12 gain stages in 6dB steps. The front panel also offers phantom power, phase reverse, and a push button for switching the unit between 62 and

The minimal layout on the cigar case begins with a white ‘on’ LED, a mini USB port for computer connection, a stereo ¼-inch headphone jack (which also serves as the main stereo out) and a rubberised volume pot mid body, which has a surprisingly reassuring, professional feel for such a small pot. If you’re looking for a high quality Class-A digitalto-analogue converter for listening to your music library – or any other sources such as films etc – you may need look no further than the American built Centrance DACport. Looking like a single aluminium cigar case, the 24-bit/96k-capable DACport has a very small footprint that allows it to be hidden behind your amplification setup or simply plonked beside your computer’s USB port. The 11cm long DACport even comes with an aluminium clip (that’s presumably designed to allow you to attach the converter to your jeans while you’re on the move), but unfortunately this quirky option is ludicrously loose and is all but useless as a holster for the device. It does, however, work well as a stand for the device, lifting it up off any precious wood surfaces it may sit on. Being a Class-A device the DACport is quite warm to the

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The DACport uses Centrance’s proprietory ‘Adaptiwave’ technology, making the unit driverless and therefore truly plug ‘n’ play. To give you an idea of Centrance’s cudos amongst developers, this technology is licensed by the likes of Benchmark and Lavry – amongst others – for its professional data streaming technique. The technology allows the user to plug a device using ‘Adaptiwave’ into any computer running Mac, Windows or Linux, providing high USB bandwidth, all with the device seamlessly appearing in the list of sound devices. Select your device – in our case the Centrance DACport – and you’re ready to go. There are no configuration or mixer windows, driver panels or other hullabaloo. Similar to other Class-A devices, the DACport takes a few minutes to hit its optimum sonic sweet spot but when it does the sonic improvement over my onboard

How’s it sound though? I’ll be honest. This isn’t a vintage V76, but it does sound very good (the V76 was never a solid state device anyway). But yes, it certainly has been inspired by the V76 and preamps of its ilk – clear yet characterful. Being such a highgain unit it definitely excels on my cheap and notso-cheap ribbon mics, which is why I won’t be letting this preamp out of my grasp. For the money, I believe you’re looking at a real bargain. Brad Watts. Price: $609. Professional Audio Services: (02) 6059 1652 or

D/A conversion is staggering. Indeed, the DACport even stacks up well against my RME Fireface converters and is a worthy contender in a shootout with my trusty UA 2192, only pipped at the post by the UA’s fantastic super-low bottom-end presence and slightly higher headroom (not bad for a cigar-sized converter with a price tag around $400). One of the characteristics I liked most about the DACport was the uncluttered yet solid lower mids, providing a ‘see through’ definition without being wimpy between 150 and 800Hz – a trait usually reserved for the high-end converter market frequented by the likes of Crane Song, Apogee, Lavry etc. Another nice sonic feature is the extended top end that’s never overbearing or too open and silky sounding. Overall I think the DACport is a really classy converter that will provide hours and hours of excellent non-fatiguing listening. The only downside I can see is the DACport’s lack of input converters or the fact that you’ll have to buy or solder up some leads for your designated output device. These minor hitches are nevertheless a small price to pay for making your digital audio library or gig laptop sound this good. Cigar anyone? Calum Orr Price: US$399

Oh red and beautiful Nord Stage 2 Nothing in the world compares to you Handmade by devoted minussweden at clavia dmiicablovers

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ZOOM Q3HD HANDY VIDEO RECORDER which I recently used on my travels in Fiji. I recorded two albums ‘in the field’ with the device, getting to know the ins and outs of it in a classic baptism of fire. In short, the unit sounds great, although the Q3HD is predominantly geared towards shooting HD video – which it does in wonderful widescreen 1080P HD. Nuts! Upon turning the unit on, the Q3HD understandably defaults to Video mode (the thing runs for hours with two AA batteries by the way, depending on whether you’re capturing video and the nature of the batteries). Changing this default setting to audio-only capture involves a trip to the menu page, which is simple enough – the menu button is dead centre on the unit, just above the record button. Problem is, if you don’t then initiate an audio recording within two minutes, the unit turns off again. Frustratingly, the next time you turn the unit back on you’re back in video capture mode. This prompts another trip down menu lane to the ‘Camera’ icon and the ‘on/off’ option – annoying, but not life threatening. It’s just a shame there’s no way to permanently alter the default boot setting.

Seriously, when I think about how expensive, fragile and temperamental early portable two-track DAT recorders were compared to the current crop of inexpensive, reliable and easy-to-use digital solidstate devices, I feel like the money spent all those years ago would have been better gambled down the dog track. The latest of these devices to rub salt into my old wounds is the Zoom Q3HD Handy Video Recorder,

The audio captured to the SD (or SDHC) card happens via two relatively cheap but reasonable sounding in-built condensers permanently arranged behind a sturdy plastic grille in a 120º X/Y configuration. I had no problem with the sound of these mics, more an issue with their stereo preamp, which only has three settings: low, high and automatic. Automatic, it turns out, works quite well, but I’d much prefer to have had the ability to dial up a level with a pot or wheel. There’s an alternative to this: a line-in on mini-jack… not exactly a professional option, but an option

nonetheless. The unit also features a tiny speaker (for monitoring 1kHz!) and a low-cut filter for the inbuilt mics, available via the main menu. This is handy for minimising handling noise and rumble… both of which it’s quite prone to. It’s best to use the device on a tripod, for which there’s a screw mount in the base. Down the side of the unit, alongside the gain switch are a headphone/line output, an HDMI output and a ‘TV’ output, also on mini-jack. The beauty of the Q3HD is its utter simplicity, and cost. It’s a bargain at under $400 (street price), and frankly, two wouldn’t go astray for field recording duties if you could afford it – one for close-up detail, the other for ambience. A perfect example of how practical it is in the field was highlighted immediately after I concluded my first recording session. I adjourned to my laptop to download the 24-bit/48kHz files (there are other resolutions: PCM 44.1, 48 and 96kHz at either 16- or 24-bit, or AAC 160 to 320 kbps at 48kHz) and suffered a momentary panic attack as I scratched around for a mini-USB connector. Then I remembered the one-inch long USB cable and connector tucked into its side. Problem solved. There are literally no leads required! Perfect. For the price, this is another great two-track digital field recorder/no-brainer from Zoom. I can’t think of many reasons why every audio engineer shouldn’t have one. Andy Stewart Price: $469 Dynamic Music: (02) 9939 1299 or

RØDE VIDEOMIC PRO high-quality HDSLR cameras hitting the streets, a revolution in multimedia production and consumption has taken place, and is spreading like an ash cloud. Soon there will be Full HD motion picture cameras in every middle-class home in the world, and the Rode VideoMic Pro is designed to fit neatly on top of every one of them.

Røde Microphones has an impressive array of shotgun mics in its range these days. The latest of these is the VideoMic Pro, a mini-shotgun mono condenser, released a few months ago now and aimed fairly and squarely at the video capture, on-camera microphone market. With the recent explosion of relatively low-cost and extraordinarily

The VideoMic Pro is a balanced sounding, superlightweight, neatly designed and modestly proportioned electret condenser. Its power is derived from a nine-volt battery, which tucks into the base of the mic, inside the nifty cradle shockmount. Despite appearances, the mic does not draw power from a camera’s hot-shoe when it’s mounted in this fashion, but it does accept a screw thread for connecting to a boom pole. There’s an 80Hz high-pass filter for minimising rumble from the surrounding environment or handling noise, and a three-position level switch (–10dB, 0dB and +20-dB) for basic level adjustment. At only 85g and measuring six inches in length, the VideoMic Pro is the perfect fit for these physically smaller cameras. The VideoMic Pro will appeal to a wide cross-section

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of the professional/enthusiast videographer market – ‘pro’ enough for the pros, and ‘simple enough to use’ for the countless thousands of video enthusiasts out there. One thing to note: some people might be a little confused about why there’s a stereo mini-jack output fixed to the body of the mic when the mic itself is mono. The stereo mini-jack feeds a dual-mono signal to the left and right-hand sides of a stereo mini-jack input. The mic is most certainly mono. This prevents that annoying issue of the signal appearing on only the left or right side of headphones during a recording session, or striped to only one side of stereo two-track. If you’re one of the legions of videographers out there looking for an on-camera mic, Rode’s VideoMic Pro is a competitively priced and simple to use option. It also comes with a 10-year warranty. Hard to go past really. Andy Stewart Price: $299 Røde Microphones: (02) 9648 5855 or

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QUESTED V2108 & X15

The quest to discover the perfect nearfield continues. Text: Andy Stewart

Staring into a pair of the new flagship Quested V2108 two-way active monitors is an imposing, almost mesmerising sight. Under certain lighting conditions they appear as virtual Black Holes, mixes seemingly emanating from hyperspace rather than a pair of three-dimensional objects. Apart from their (mercifully) low-key blue LED power indicator light and Quested logo on the cabinet’s front panel, these speakers would definitely get my vote for ‘blackest speaker on the market’.

a tape measure to both they’re actually identical: 337mm x 400mm x 340mm. The difference is simply that the heat sink on the new model, which cools the significantly more powerful twin MC2 Audio Class-AB power amps, extends the cabinet’s overall depth by 25mm. There are significant differences, not least of which being these new amps, but the changes are arguably more compelling than superficial similarities would first seem to indicate.

Seriously though, this latest offering from Britain’s king of the soft dome tweeter, Roger Quested, is surprisingly different sounding when compared to its older brother, the similarly named and constructed VS2108A, released to great applause back in 2006. Time to investigate why…

The new V2108 is unquestionably a more balanced sounding speaker than its ‘VS’ predecessor. The bass response is far more self-assured and extended – down 2dB at 40Hz; the previous model dropped to around –7dB at 40Hz – and the midrange seemingly less strident between around 2 and 4kHz. I’m not certain whether this is simply a difference between the two pairs of speakers I have in front of me – especially given my VS2108As are about six years old now – or typical of the differences generally, but my preference is certainly for the new model. It’s claimed by Roger himself that the V2108s represent a ‘ground-up’ redesign, but the cosmetic similarities between the models are so compelling my eyes have been working overtime to resist this assertion. The main question in my head has repeatedly been: ‘how can this be a ground-up redesign when the boxes are seemingly identical?’ But in the end sound is what speakers are all about, and if you close your eyes and A/B the two models there’s no doubting the difference.


Although both models appear quite similar on the surface, apart from minor cosmetic issues – the doping on the driver cone is now blue/black, not green and the Quested logo is blue/grey, not pink – the obvious visible difference is that the dual bass ports of the V2108 are positioned at the bottom rather than the top of the front panel. When I first saw this new configuration of eight-inch driver flanked at the corners by bass ports I thought it looked slightly odd. Now that I’ve gotten used to it, suddenly it’s my venerable VS2108As that look out of whack, their high riding bass ports making the front panel look like a child’s portrait of Mickey Mouse, an image I can’t now seem to get out of my head. The other theoretical difference with the Quested V2108 (according to the manual at least) is that the cabinet of the new model is slightly larger. However, when you take AT 64



To give the new speakers some real-world context, in recent times my own pair of Quested VS2108As have played

Overall, the new model sounds stronger, flatter, and more balanced across its more expansive audio spectrum.

second fiddle to my Event Opals mainly because, by comparison, the Questeds are more strident in the midrange, far less powerful, and less expressive in the low end. The new Quested V2108 positions itself somewhere between these two options – not quite as responsive down low as the Opals, nor as powerful, but offering more midrange definition without tipping over the edge into a world of harshness and fatigue. They’re also more powerful than the older Quested model: the VS2108A was always underpowered, its driver serviced by a 110W RMS amplifier, whereas the driver and tweeter of the new model enjoy the fresh impetus of two independent 200W amplifiers that were custom built specifically for Quested by its affiliated company, MC2 Audio, making the speakers more effortless sounding and louder overall by 4dB. The other issue I’ve always found a little troubling with my VS2108As is that they seem to have a ‘hole’ in them right smack in the middle of the frequency response – lacking strength between around 900Hz and 2kHz, and over-compensating for it just above this region, This makes them slightly scooped and hard sounding the moment any mix had even the slightest excess of upper midrange content. This is not the case with the V2108s. Overall, the new model sounds stronger, flatter, and more balanced across its more expansive audio spectrum. One notable physical difference that goes some way to accounting for this tonal shift in the re-engineered design is the all-important active crossover point between the driver and tweeter, which has been elevated slightly from 1.25kHz to 1.4kHz, and doubled in gradient, from 12dB per-octave to 24. This seems to have helped cure the cabinet of its midrange ‘hole’, though I’ve not seen a frequency response plot of the new model to confirm this conjecture. (It’s conspicuously absent from the decidedly homespun looking, inkimpoverished manual.) One thing’s for sure, the soundstage produced by a pair of the new V2108s is impressive; the slightly stiffer eight-inch driver and softdome tweeter reproducing room ambience and spatial cues with great detail yet without hype. More so than the previous model (which I’ve been using for six years), these speakers are one of those genuine listening tools that neither flatter nor undermine an in-progress mix. As a mix comes together the speakers begin to light up; good work consistently being rewarded by an improvement in the image and tonal balance until finally the results can be quite stunning, assuming you’ve succeeded in your work. If you haven’t – if you’ve pushed the midrange too far they sound harsh; if there’s not enough bass they sound thin; if it’s too slammed they sound choked, and so on. Unlike older speaker standards like the ubiquitous Yamaha NS-10M, for example, a finished mix that features a good balance of frequencies and three-dimensionality will never fail to impress on the V2108s. NS-10s rarely do. Like so many speakers on the market these days, the V2108s have the high and low contour switches, and input sensitivity controls

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The overall combination of this 2.1 system is to die for. Mixes sound truly enormous, in-progress work becomes a pleasure and no frequency is left untuned (at one stage, while working up a kick drum sound, I almost became convinced my head was literally inside the drum, so compelling was the sense of immersion and scale). The system can run very loud too and holds together nicely, although I’d wager it wouldn’t sound as impressive in a smaller control room. This system needs a fairly large room in which to breathe.

The X15 sub is a passive speaker that works seamlessly in conjunction with the sophisticated SBC800 powered sub-bass controller.

NEED TO KNOW Price V2108: $2999 each. X15 Sub: $2999. SBC800 controller/amp: $2999. Contact CMI (03) 9315 2244 Pros Tonally balanced and powerful sound. Unassuming good looks. Vastly improved onboard amplifiers. Great imaging and dynamics. Black. Cons Fit & finish on the sub looks a little homespun. Some may find the Quested aesthetic a little dated. Summary Studio monitors are a personal thing, and the key to getting them right is knowing them backwards in your own space. I’ve used Questeds in umpteen commercial and home studio spaces over the years, and never once have I found them to be anything but impressive. The new V2108s are every bit as good as previous models, and in other respects, even better. Put simply, I wouldn’t suggest you go to the hassle of trialling them in your own place if I didn’t think they were any good!

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onboard should you need to adjust the speakers to better fit your acoustic environment. Unlike the last time I reviewed Quested monitors – the V3110s – this time around I didn’t have to adjust the controls at all to get the sound I was after. The recessed HF contour dip-switches offer stepped adjustment of ±1dB shelving at 2.5kHz, while the LF ‘compensation’ control, which requires a flat-head screwdriver to adjust, provides four options: –9, –3, Flat or +1dB at 40Hz, the first of these settings being designed for matching the V2108s with a sub. THEN ALONG CAME A SUB

There have been two other significant components in my possession during the couple of months I’ve worked with the V2108s, one of which I found hard to get through the door when it first arrived! These are the passive Quested X15 sub-bass speaker cabinet and SBC800 powered sub-bass controller. In combination with the V2108s, the 15-inch passive speaker and associated management system will almost certainly have hoons pulling up at your doorstep looking for that hidden street machine or doof party. Set up as a nearfield 2.1 system on my Neve console, with the X15 placed behind it and the 2U sub-bass controller placed literally in it, I suddenly found myself immersed in my mixes rather than seated in front of them. Having comprehensive control over the X15 via the SBC800 amplifier (and a flat-head screwdriver) made all the difference. In the same way as a recording/mixing console is more effective as a tool when the knobs and faders are on the top of the console rather than underneath it, having the sub-bass speaker management system beside you rather than buried underneath or behind an oafish 15-inch speaker cabinet makes refining the balance of the system a breeze. Instead of having to run back and forth between the sub and the mix position to make crude adjustments based on (at best) an educated guess, or worse, a hunch, having the controller beside me while I worked was the difference between trusting the sub implicitly and playing a dubious guessing game. Being able to adjust the gain, phase (right through 180º), and crossover (between 30 and 90Hz), as well as switch the sub on and off altogether or solo it, resolved any misgivings I might have had with the system. With the controller placed in easy reach, I was easily able to adjust the system repeatedly over several sessions until I was fully satisfied with the balance.

There is one criticism I must level before I close proceedings here. As is the case to a much lesser extent with the V2108s, the X15 sub feels a little ‘homemade’ in terms of the cabinet’s construction and finish. Where so many other companies – Genelec, Event and ATC etc – have gone to great lengths to develop sophisticated enclosures featuring advanced ports and non-parallel internal structures made of metal alloys etc, rather than wood, Quested remains staunchly embedded in the ‘square wooden box’ camp. This is not a criticism of the wooden box design principle as a whole I must stress, but rather an observation about the finish of the Quested cabinets generally. Personally I like the look of the V2108s – they’re sturdy, unassuming and powerful looking – and looks do matter when you have to stare at them 12 hours a day – but the X15 sub in particular looks a bit like it’s been crafted out of an old Rank Arena TV enclosure, with the front ‘grille’ made by simply carving slits into the structural timber with a jigsaw. It looks a bit ‘rough’, shall we say, and not in keeping with how impressive it sounds, which is a shame. The earlier model VS1112 active sub, which I currently use in combination with my VS2108As, looks far more refined by comparison. These cosmetic issues aside, the sound of the V2108s is very impressive, and in combination with the X15 sub and SBC800 management system/amplifier, even more so. The system retails for about 12 grand though, which is a huge pile of cash, although you’d obviously get it for significantly less than that if you haggled with a retailer, but I’d wager you’d be hard pressed to find a better system that so comprehensively covers the audible frequency spectrum. But of course most people aren’t looking to replay 25 – 40Hz anyway – it’s more of a luxury than a necessity. Given that, on their own, without the sub, the V2108s sound fantastic and I wouldn’t hesitate for a moment to recommend them to anyone who’s toying with the idea of trialling a pair. Quested monitors are a brand of speaker I’ve used successfully for years so I’m perhaps a little biased. In the end the best studio monitors are the ones that satisfy you for long enough that eventually you come to know them intimately. That’s been my experience with a small handful of speakers over the years, Quested VS2108As being among them. If you’ve often been curious about the sound of Questeds but not worked with them before, now is arguably the best time in a decade to investigate whether their sound might be right for you. In the end only your ears can be the judge, of course, so haul them (along with the rest of your body) to wherever you can check them out.

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Very few of us like to conform to standards and norms, but when it comes to LKFS scale you probably should. Text: Brent Heber

It’s the summer holidays, you’re sitting around relaxing with relatives and it somehow gets mentioned that you’re a sound engineer. Next thing you know you’re being held personally responsible for all the deafening and irritating television commercials and forced to endure a tirade from the seniors in the group (who ironically should possess less acute hearing and be less worried by the trend). Why do we have to endure this persecution? After all, we don’t like dynamic range compression any more than they do, but it’s a necessary evil, right? What other ways are there to monitor and control levels without human intervention? There’s been a shift in thinking on this topic over the last few years and new software tools and systems have been developed and refined that help with the problem of momentary peak levels not truly reflecting our perception of overall loudness. It’s all about measuring sound signals differently... MEASURING PERCEPTUAL ENERGY

Loudness is all about measuring the perceptual energy of sound, as opposed to quantifying its purely peak or RMS-measured electrical level as a signal. Instead of measuring it at any instant in time, loudness is calculated over an appropriate period of time – more analogous to how the human ear and brain interpret sound. In a purely technical sense, loudness is quantified using the LKFS (Loudness, K-weighted, Full Scale) metering scale – a complex algorithm (specified in ITU standard BS.1770) that measures electrical energy, interprets it as audio data and monitors specific frequency bands over time. This then relates such a measurement to how the ear behaves, and comes up with a value (ongoing refinements to these AT 68

algorithms are continuing that will hopefully lead to the adoption of a Loudness Units and LUFS scale). So how does a broadcaster use these LKFS measurements? They use normalisation! Let’s say your program has a wide dynamic range, like a film or some orchestral music, that is perceived to be relatively quiet most of the time. In this typical scenario the program might have a low loudness value over time – say –31LKFS. Playing straight after it is a TV commercial that’s heavily compressed and perceived to be very loud. This will have a much higher loudness level value of say –15LKFS. One of the intelligently designed aspects of this system is that the unit of measurement – 1LKFS – is roughly equivalent to 1dB. So in the example above, if the commercial is attenuated by 16dB before playback, there will be no significant level jump for the listener achieving the goal of consistent loudness on broadcast. Loudness monitoring equipment is currently being rolled out amongst broadcasters worldwide that will give them the capacity to measure program material and either normalise it before broadcast or send it back for remixing to meet their loudness specifications. KNOW YOUR LKFS

Importantly, loudness can be measured either across the whole frequency range of your program or specified in terms of Dialog Level, where measurements of loudness are only considered within the range of the human voice. If you’re delivering something that requires a pre-ordained LKFS level, be sure to check which of these two types of measurement you need to mix to – generally advertising material is measured across the whole program while longer form material is typically determined with respect to Dialog Level loudness.

The TC Electronic’s ‘spin’ on the LKFS meter – the LM5D plug-in – not only includes the true peak and LKFS information but presents its results as a visual graph over time, revolving in the centre of the GUI.

If you’re mixing for TV, do you know if your program is going to be analysed with respect to its loudness? Will the broadcaster normalise your material to adhere to its loudness specification or will it expect you to deliver it at the right LKFS value? (In Australia, a standard for digital transmission on the FreeTV network was adopted last year specifying average loudness at –24LKFS for broadcast.) Depending on the job, you may need some loudness meters to check your mix before handing it over.

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So where do you find a meter that measures in LKFS? In ProTools perhaps? No. Don’t bother opening the standard ProTools Phasescope; it’s not there as a ballistics option. The closest thing you will find to it is ‘LeqA’, which is an early precursor to LKFS. The good news for producers working in the box is that there are now quite a few LKFS metering options, some of which are very cost effective. Also, don’t panic about changing the way you work – due to the relationship between the decibel scale and the LKFS scale, you can mix and produce the way you always have. When you finish mixing your project, simply measure its LKFS value in software and normalise the program up – or more likely down – by the required dBFS trim to achieve the magic –24LKFS number. DOLBY MEDIA METER

The first loudness metering plug-in on the market was developed by the company close to the heart of the technology, Dolby. The Dolby Media Meter (now in version 2) measures whole program loudness, dialog level, loudness range, long-term vs short-term values, and is easy to configure visually. With graphing functionality you can see exactly where your program is triggering overs, and work both in real-time and offline – the offline analysis is achieved either using the Audiosuite version of the plugin in ProTools or a standalone version with other DAWs.

Benefits • Fits any large-diaphragm recording microphone with a diameter between 18 & 55mm • Complete Pop Filter & Suspension System • USM offers up to 12dB better isolation against unwanted vibrations

“It’s the BEST shock mount ever made! This mount is tr uly awesome. This thing is incredible and works on ever ything”. Fred Coury – Drummer/Engineer Producer For your nearest authorised dealer in Australia: Free Call 1800 648 628 | email New Zealand: Free Call 0800 100 755 | email What flavour do you prefer?: Nugen Audio provides two flavours of VIsLM: one with data logging and one without – available across all platforms (RTAS, VST, and AU). AT 69

The Dolby Media Meter V2: The first loudness metering plug-in on the market. Now in v2, the DMM measures whole program loudness, dialog level, loudness range, and long-term vs short-term values.

“ ”

What other ways are there to monitor and control levels without human intervention?


Another great loudness meter is produced by TC Electronic – the LM5D plug-in. This meter runs on ProTools HD (unfortunately there’s no native version as yet) and not only includes the true peak and LKFS information but presents its results as a visual graph over time, revolving in the centre of the GUI. The LM5D ‘radar’ meter is also at the heart of TC Electronic’s loudness monitoring hardware, the TC Touchmonitor TM7 and TM9 and the LM2 Stereo Loudness Meter. The LM5D also displays its measurements to within 0.1dB, a useful difference, but will only measure in real-time due to DSP usage. NUGEN AUDIO

Moving to the more affordable end of the spectrum, Nugen Audio provides two flavours of its VIsLM, one with data logging and one without. Importantly, it’s also available across all platforms (RTAS, VST, and AU) like its more expensive counterparts. The team at Nugen Audio have been closely involved with the EBU, even co-presenting their solutions on the official EBU stand at the recent IBC trade show. Their plug-ins have been developing rapidly over the last year and the team is quick to adopt changes to specifications as and when they occur. They also support Audiosuite file analysis like the Dolby Media Meter. VSONIC & AUDIOLEAK

Another option on a budget is VSonic’s ‘VMeters’, also available on all platforms. Interestingly, VMeters is the only option that not only provides loudness information but also meters configurable to the Bob Katz ‘K-Metering’ standard (worth reading up on if you have a spare few minutes). The third budget solution is the Mac-based AU tool ‘Audioleak’ from Channel D. There are actually two versions of Audioleak that measure LKFS, the difference being support for 5.1 measurements. Although Audioleak doesn’t work natively inside ProTools, it’s a simple matter to drag and drop files onto it for analysis. No leaks here!: The Mac-based AU ‘Audioleak’ from Channel D comes in two versions, one of which supports 5.1 measurements. It doesn’t work natively inside ProTools – simply drag and drop files onto it for analysis.

V Sonics’ ‘VMeters’ are available on all platforms and is the only option that not only provides loudness information but also meters configurable to the Bob Katz ‘K-Metering’ standard. AT 70

So whether you’re delivering a small slice of program involving these specifications once in a blue moon, or fully immersed in the cut and thrust of broadcast audio day in and day out, there are plenty of tools available to help you grapple with the new loudness measurements. Obviously a big influence on your mix is the desired instantaneous peak level of your program contending with the loudness over time. The balance between the two will largely dictate your compression strategy and consequently the dynamic range of your program. The tools listed above will all provide you with what you want to know and when you want to know it. Fingers crossed, the wider audio community will embrace an arguably smarter way of measuring our perception of sound and, in turn, gradually move us beyond these dark days of the Loudness War.


Dolby Media Meter v2: $972 (Syntec International: 1800 648 628)

TC Electronic LM5D: $1499 (Amber Technology: 1800 251 367)

VIsLM: US$299 without logging, US$449 full version. (

VMeters: US$79 stereo only; US$129 surround-capable. (

Audioleak: US$36 standard license, US$137 pro version. (

Pros Many options that scale according to usage and budget. Measuring loudness rather than peak means less compression is needed to hit the target values. Over-compressed material gets pulled down, so maximising is no longer desirable. Cons Some of the plug-ins only support real-time measurement. Summary Measuring sound the way the ear perceives it could be the industry shift we desperately need to save us from overcompression and ear fatigue. It will also save us from those angry relatives.

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With no ports, no built-in amplifiers and a preference for passive crossovers, Grover Notting takes studio monitoring into the future by delving into the past… Text: Greg Simmons



Further information related to this review, including an interview with Grover Notting’s founder Frank Hinton about the development of the Principal Monitor series, plus commentary by Dr Neville Thiele and Graeme Huon about loudspeaker enclosures, filter design and the Bandwidth Extension Module, can be found on the AT website.

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When Frank Hinton of Grover Notting told me he was designing a series of principal studio monitors, I was outwardly encouraging but secretly cynical. It was September 2009 and Grover Notting had been enjoying some success with their CR series of cross-reference monitors [reviewed by Andy Stewart in Issue 65]. In fact, I’d recently bought myself a pair of CR1s after using them to cross-reference some recordings I’d done for a Josh Pyke DVD. I was impressed by their stereo imaging, their easy-going nature compared to the nagging NS10s that sat beside them, and by how well the results translated to the outside world. The CR series were successful because they filled the hole left in the market after Auratone’s 5C and Yamaha’s NS10 had faded into irrelevance. I could see no such hole in the principal monitor market, however, and felt that another series of principal monitors was precisely what the world didn’t need. I listened politely nonetheless, eyebrows raised to 50% in half interest, as Frank enthused over his forthcoming ‘Principal Monitor’ series. “These new monitors build on the Audio Information Band concept we developed for the CR series,” Frank began. Grover Notting defines the ‘Audio Information Band’ as the extremes of the human vocal bandwidth, from 80Hz to 11kHz, and believes that a useful monitor must be able to reproduce that bandwidth with a linear amplitude and phase response. I raised my eyebrows to 60% out of respect for the CR series. “All these new monitors use infinite baffle enclosures – no ports,” he declared. Interesting… I had always preferred the clearly defined low frequency performance of an

infinite baffle enclosure to the more extended but woollier ‘thump factor’ provided by most of the ported enclosures that dominate the market. My eyebrows involuntarily notched up to 70%. “There are no amplifiers bolted onto their backs,” Frank continued. Eyebrows to 80%. When active monitors first hit the market I loved the idea of integrating the amplifier into the speaker enclosure, but I had since grown to dislike the added weight and cabling. I yearned for the days when amplifiers lived in ventilated racks where they belonged, and monitors could be moved around without slipping a disc or popping a hernia. “They’re all using passive crossovers designed by Dr Neville Thiele…” Eyebrows at 90%. Dr Neville Thiele is one of the world’s leading experts on the design of loudspeaker enclosures and filters. In 1961 he and Dr Richard H. Small published their famous paper, Loudspeakers in Vented Boxes, replacing all the guesswork of vented (i.e. ported) enclosure design with a methodical approach backed by solid mathematical formulae. The ‘Thiele/Small parameters’ are now a fundamental part of loudspeaker enclosure design. Frank’s enthusiasm was contagious, but I still had doubts. How could an infinite baffle enclosure compete in a market dominated by ported enclosures that, on a size-forsize basis, produced more low frequency output? If only there was a way to combine the low frequency clarity of the infinite baffle enclosure with the extended bandwidth of a ported enclosure.

With their infinite baffle designs, high quality passive crossovers, lack of built-in amplifiers and the Bandwidth Extension Module, Grover Notting has ignored every market trend and taken the design of studio monitors back to the drawing board

“They’re also the first loudspeakers in the world to use Graeme Huon’s patented Bandwidth Extension Module,” Frank concluded, clasping his hands. “Think of it as a passive electronic version of a port. It gives an infinite baffle enclosure the extended bandwidth of a ported enclosure, without any of the acoustical problems inherent in ports.” My eyebrows ran out of headroom at this point. By now Frank had my undivided attention, and I hope I’ve still got yours…


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Now in production, Grover Notting’s Principal Monitor series consists of six models intended for applications from desktop mixing to large scale mastering. Proudly designed and manufactured in Australia, the series represents a complete re-think of critical monitor design with no ports, no builtin amplifiers, no active crossovers [except for the models incorporating sub-bass drivers] and a strict adherence to the Audio Information Band concept. Every design decision was made to achieve the highest performance, rather than meeting a price point or following a market trend. They’re not cheap, but if you’re looking for a no-compromise studio monitor that was designed entirely in response to the needs of working audio professionals, you might find it in the Principal Monitor series. The Code 1.5, reviewed here, is the smallest in the series. It’s a passive two-way system in an infinite baffle enclosure measuring 285mm x 175mm x 164mm and weighing a satisfying 6kg. Intended for use in desktop mixing and similar near-field applications, and as a portable monitor for location work, it offers a frequency response of 75Hz to 32kHz (–3dB), a sensitivity of 85dB SPL (1 watt @ 1m), and typically less than 0.2% THD in the mid-band. The Code 1.5 uses a 149mm (a tick under six inches) mid/bass driver and a 26mm soft dome high frequency driver. The drivers are the heart of any studio monitor and these ones are worthy of their own reviews, but I don’t have the space for that here. You can read about them on Grover Notting’s website (www., and in the interview with Frank Hinton that accompanies this review on the AT website.

• Up to 150MHz tuning range • 256 bit RC4 signal encryption for secure audio transmission • 2-channel digital true-diversity receiver • No Compander (used in analogue systems): higher sound quality • On-board DSP per channel (Compressor, EQ, Limiter) • Quick setup via infrared data link to the transmitter • Graphical spectrum analyser helps find clear channels • Remote monitoring and control via PC

A passive fourth-order Linkwitz/Riley crossover, designed by Dr Neville Thiele, as mentioned earlier, provides the frequency division at a respectably high 2.5kHz. Finally, Graeme Huon’s Bandwidth Extension Module adds an extra half octave or so of low frequency performance while sharply filtering out any low frequencies that cannot be reproduced by the system, thereby easing the load on the mid/bass driver and the amplifier. Speaking of loads, the Code 1.5 has a nominal impedance of 8Ω. AT 73


I spent many hours with a pair of Code 1.5s in numerous control rooms at the Australian Institute of Music, with the primary evaluation sessions being conducted over two five-hour periods on two consecutive days. The first was a listening session to establish what they were capable of. I was using Audiofile Engineering’s Fidelia software player through my MacBook Pro’s S/PDIF optical output into a Benchmark Media Systems DAC1 driving a Bryston 4B power amplifier. The test material ranged from 16-bit/44.1kHz wavs lifted from my CD collection to my 24-bit/96kHz field recordings from Asia.

NEED TO KNOW Price Code 1.5: $2379 each. Contact Classic Audio Designs (03) 9379 5025 Pros Highly accurate without sounding clinical. Tonal consistency and naturalness. Excellent low frequency resolution. Remarkable soundstaging. Solid construction. Designed, tested and manufactured in Australia. Cons Low SPL capability and limited low frequency extension due to their small size. Passive design means the price does not include power amplifiers. Summary The Code 1.5 is designed to meet a performance level dictated by audio practitioners, not a price point dictated by a marketing department. It’s fair to say that it is expensive, but not fair to say that it is too expensive for what it offers. Its small size dictates a limited SPL capability and reduced low frequency extension, but within those limitations it is uniquely accurate, natural and useful. It would be very difficult to find a monitor of similar size that offers equivalent performance.

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I’ll start by saying that with a –3dB point of 75Hz, a sensitivity of 85dB SPL and a typical SPL capability of about 100dB SPL, you won’t be partying to the Code 1.5s. However, within their SPL and low frequency capabilities I cannot find anything to fault, nor to ask more of; there are no prominent peaks or dips, no nagging colourations, no power compression and none of the ‘small box’ effect I was expecting. The balance of parameters is, dare I say, perfect; the Code 1.5s offer one of the most accurate and rewarding listening experiences I’ve ever had. Instead of prattling on about their highly linear amplitude and phase responses, their unfatiguing low distortion and all the usual stuff you read in monitor reviews, I’m going to focus on the one telltale performance aspect that only works well when all of the other factors are taken care of: soundstaging, i.e. a sense of width and depth, along with clearly focused image location. For this, the Code 1.5s are truly remarkable. Close-miked popular music was revealed in all of its two-dimensional in-your-face glory; you could almost see the wall of sound confined between the monitors and extending back no more than 30cm or so behind the baffles. Meanwhile, all-tube direct-to-stereo acoustic recordings from Water Lily Acoustics and challenging solid-state acoustic recordings from ECM both created three-dimensional soundstages that extended beyond the monitors in all directions. I wanted to take a walk inside… The quality of the low frequency reproduction is particularly unique for a monitor of this size. It drops off quite sharply and noticeably below 75Hz, but all the way down to that point it is as linear, detailed and controlled as I’ve heard through any monitor of any size. I attribute this to Graeme Huon’s innovative Bandwidth Extension Module. If my experiences with an early prototype version of the larger Code 4 are any indication, it’s safe to say that the larger models in the Principal Monitor series, with their extended low frequency responses, will perform even better. WALKING THE TALK

It’s one thing to listen to a pair of monitors, but quite another to work with them – that’s what the second evaluation session was for. I used the same monitoring path but ran some troublesome mixes of my own from within Apple’s Logic Pro. These mixes had been done on a range of active two-way nearfield monitors in ported enclosures from Genelec, Dynaudio, Mackie and similar; some of them I had re-visited and tweaked on numerous occasions. I knew they weren’t right but could never put my finger on the cause of the problems. The Code 1.5s laid those problems out in front of me as if they were the most obvious things in the world. Panning, equalisation, dynamic processing and reverberation problems were a cinch to identify and fix. Many of the problems were due

to subtle low frequency masking that the ported monitors I’d previously mixed on could not reveal due to their imprecise low frequency behaviour, but which the Code 1.5’s infinite baffle enclosure and Bandwidth Extension Module reproduced cleanly and in perspective. The combination of low frequency resolution and remarkable soundstaging made the Code 1.5s a pleasure to work with. The speed and confidence with which I was able to isolate and remedy mix problems should not be undervalued, as anyone working professionally will readily appreciate. TONAL UNIFICATION

After spending the last two decades being force-fed the benefits of active designs, I am still finding it hard to accept that the Code 1.5 is a passive design. It is clear to me that Dr Neville Thiele’s passive crossover is at least the equal of the best active designs, possibly superior, and may be the catalyst for my next observation. There is a ‘naturalness’ and tonal consistency in the Code 1.5 that I cannot remember hearing in any active monitor – and I’ve heard quite a few. I suspect this tonal consistency is because the entire bandwidth of the signal is passing through a single power amplifier before being filtered and applied to the drivers, and therefore has the tonal character of a single power amplifier embedded into it. In comparison, an active monitor filters the signal first and passes each filtered bandwidth through a separate power amplifier (each optimised for the bandwidth it is amplifying, and therefore each is different) before reaching the drivers, resulting in a different tonal character being embedded into each bandwidth. I believe that explains what I have heard in most active systems; they are capable of excellent clarity and detail, but there is always a tonal disconnect between the different bandwidths that ultimately leads to adjectives such as ‘electronic’, ‘clinical’ and ‘surgical’. The Code 1.5 has none of that disconnect; it offers remarkable clarity and detail while remaining tonally unified. NOT SO BAFFLING

With their infinite baffle designs, high quality passive crossovers, lack of built-in amplifiers and the Bandwidth Extension Module, Grover Notting has ignored every market trend and taken the design of studio monitors back to the drawing board. As a result, it’s produced a series of superbly accurate and useable studio monitors that make their active and ported competitors look more concerned with meeting the needs of the marketing department than the needs of the market itself. The Code 1.5 is the smallest member of the Principal Monitor series, and is without peer for its size. I cannot think of a monitor of superior adeptness at any size, assuming a level playing field with the source material and playback conditions kept within the Code 1.5’s SPL and low frequency capabilities. Many users will find it hard to justify the price for such a small monitor – especially those working in genres that consistently contain information below 75Hz, and who would be better served by the larger Code 4 and Code 5 models. If, however, you need uncompromising sound quality from a very small enclosure, you’ll have a hard time finding anything superior to the Code 1.5. It’s bound to raise a few eyebrows…


DISTRIBUTED BY: Musiclink Australia, 29 South Corporate Ave. Rowville VIC T: 03 9765 6565 F: 03 9765 6566 W: E:



Clockwise from Bottom Left: The board featuring the JLM99V Class-A op amp; laying out the ingredients; the preamp close to completion; and Guy Harrison looking thrilled with the finished product!

We all love a quality product, especially one you can make yourself. Text: Guy Harrison

JLM Audio in Brisbane has been servicing and supplying equipment to the audio industry for many years now. Indeed, there are precious few studios in Queensland that don’t own a piece of JLM gear. With a long history in equipment repair, there are very few faulty audio devices that haven’t landed on Joe Malone’s workbench at one point or another. Armed with the experience of having brought so much audio equipment back to life, Joe has created his own brand of extremely popular DIY kits that follow some of these classic design principles, albeit with design additions and modifications of his own added to the mix. Combining modern components and a contemporary appreciation for issues of impedance and connectivity with traditional circuit designs, JLM Audio has created some very cost effective, high quality pro audio equipment that is championed by engineers and musicians all over the globe. If you’ve ever wanted a quality piece of audio equipment that’s designed to a standard, not a price point; if you’ve been chasing that vintage sound but lack the requisite vintage budget; if you’ve got some basic soldering skills but have always thought building a circuit was beyond you, then hopefully this article will help launch your DIY audio career.

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JLM Audio kits are many and varied, ranging from simple VU meter kits through to more complex Opto Compressors and multiband EQs. Today we’re building the JLM Mono mic pre for review. Suppled in a cast aluminium Jiffy box with silk screening, the circuit board and components supplied in the JLM Mono kit are accessed by four screws on the base of the box. All controls and switches are on the top of the case. These include: power on/off, phantom power, pad and phase reverse switches. In the centre we find a large (and funky) chicken-head knob that controls gain; below this a smaller knob for impedance matching, XLR in and out, and a connector for DC input. The JLM Mono is a transformer balanced mic preamp with an electronic balanced output. It’s tough and rugged, and equally at home in the studio or on location – for servicing the latter, two 9V batteries (or NiMH rechargeable batteries) fit inside the aluminium enclosure, supplying 18V phantom power when required. If AC power is available a JLM SMPS (wallwart-style) power supply drives the preamp – and charges the NiMH batteries in circuit if fitted – while simultaneously boosting phantom power to 48V. The other advantage of being connected to the AC juice is that you’ll

enjoy increased headroom on the pre. The JLM Mono is designed to run on a Burr Brown Dual OPA2604AP op-amp as standard. Alternatively, you can opt to fit two JLM 99V (or comparable footprint) op-amps, though in this latter configuration the battery power option is lost. As a bonus you can swap between the two different op-amp types with no component changes, effectively giving you two or more audio flavours from the one box – simply plug-‘n’-play. Nifty indeed! Additionally, there’s a choice of two input transformers – these are soldered to the board however, so there’s no plug ‘n’ play option here, but it’s nice to have the sonic options.



Included in the JLM Mono kit is everything you need to build the preamp yourself: the aforementioned silk-screened aluminium Jiffy box, an etched single-sided PC board, input transformer, resistors and capacitors of various values, a few diodes, three potentiometers, four LEDs, four toggle switches, two chassis-mount XLR connectors, a DC input connector, two 9V battery terminals, an op-amp and socket, two knobs and a piece of Lexan that you cut to size yourself to stop the circuit board shorting on the case. For the ‘AT review preamp’ construction I’ve decided to go with both optional extras on the kit so I will be installing NiMH rechargeable batteries ($25) and also fitting the JLM 99V op-amp ($75). After laying all the parts out in a logical manner so you can get an idea of what’s what, the next step is to sort out the resistor values. Resistors are coded with coloured bands to define values – sounds straightforward, but in reality it’s not. Unless you spend a lot of time looking at resistors it’s difficult to decipher one from the other. So before construction began – as most serious electronics enthusiasts do – I checked resistor values with a meter, not just my eyes. It’s a simple matter of turning your multimeter to read resistance (Ohms), placing the probes on each end of the resistor and reading off the values. Once all your resistors are labelled – there are 13 different resistors in the Mono kit – its time to view the circuit board. COME TOGETHER

The PC board is clearly etched making it easy to see what goes where. All resistor values are marked and the first step is to get them into position. As this kit can be built with a number of different input transformers there are four resistor values and one capacitor value that vary based on the transformer you’re using. All the info for the build of this kit is available on the JLM website. It’s here where we find the info about which resistors and capacitors fill these positions. For the AT build we will be using the OEP 262A3C / VTX input transformer. A quick look on the JLM Audio website build thread reveals that for the OEP 262A3C/VTX transformer fitted to normal version Mono, we need these components: • RPad = 120R

• RGain = 68R

• Rload = 10k

• RZobel = Linik

• Czobel = 390pF

• Impedance pot 100k-log

This information also informs us that, of the two pots supplied for the impedance adjustments, the 100k-log pot is the one we’ll be using.


Large-diaphragm Studio Microphone

The MK4 is highly versatile, suitable for vocals, acoustic guitars and pianos as well as string and wind instruments. It is even great for drums! Main features include: • • • •

Large 1” gold spattered diaphragm Capsule suspension to minimise handling noise Very low self noise (10dB A) High maximum SPL of 140 dB


Armed with this knowledge we can now go ahead and place our resistors in position. Resistors have no polarity so you can install them either way around. I prefer to leave the capacitors mentioned above to one side for now, mainly because it’s much easier to solder components of the same height. The only resistor that oddly doesn’t get mentioned above is the one marked ‘Fuse’, though it’s easy enough from the pictures on the website to see that it’s 10Ω.

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Now for the diodes: unlike resistors, diodes do have a polarity, and to help get this right the circuit board is clearly labeled with a white band at one end matching the white band on the diode. Simple! Once these AT 77

Clockwise from Bottom Left: The preamp gets its first dose of power – and lights up!; the circuit board is clearly labelled so even the newbies can work it out; the legs of the gain pot get their extensions courtesy of the off-cuts from the diodes – nothing is wasted!; preamp close to completion; there are spaces on the circuit board for two op-amp options.

NEED TO KNOW Price Kit: $150 JLM 99V: $75 PSU: $49 Input Transformer: $40 - $60 Contact JLM Audio (07) 3891 2244 Pros Rugged build quality. Great sound. A bargain, provided you don’t charge yourself too much for labour. Cons No paper manual. Full headroom of the unit is only possible with ‘optional’ power supply. Summary There are very few excuses preventing genuine audio enthusiasts from building one (or several) of these Mono preamps. If the question of ‘how do they sound?’ is stopping you from attempting the build, fear not. The JLM Mono sounds great and would make a worthy addition to your preamp collection. AT 78

are soldered and trimmed up it’s time to fit the op-amp IC socket followed by the capacitors.

Lexan thermoplastic sheet we’re ready to clip in the NiMH batteries and screw up the back plate.

There are two different capacitor sizes used in the JLM Mono. The larger ones are all of the same value and flank the gain pot position on the PC board. Capacitors have polarity too: again, marked on both the PC board and the capacitors themselves. Press the caps into position and solder them up.


Toggle switches for power, pad and phase are next on the agenda followed by the XLR sockets. Once these are in position, it’s time to test the fit of the components to make sure all our toggle switches and XLRs line up correctly with the top cover. Once you’re satisfied, solder them into position. INPUT TRANNY

Next to the input transformer: this is where things get a little tricky. The input transformer has to be modified somewhat to fit onto the board and will also fit in backwards, which means you can accidentally install it the wrong way around. Clearly this is not what we want to do so take care with this step. After some considerable time spent moving the ground leg of the transformer I was able to finally solder the transformer into place. (This has since been rectified in version two of the JLM Mono, which is now available.) Next to the DC connector. The legs saved from trimming up the diodes come into play here and are used to extend the legs of the DC connector. This is a fiddly job and you’ll need a vice or PC board holder (and patience) to do the job properly. Next, solder up the LEDs and gain pot, which also get the leg extension treatment. Now is also a good time to install the smaller capacitors. Finally, after you’ve fitted up the battery terminals and impedance pot, we’re ready to slot the circuit board into the top of the case… we’re beginning to see the light at the end of the tunnel. A couple of stick-on foam pads help secure the batteries in place and after some trimming of the polycarbonate

Phew! I feel like I’ve worked for it, but she’s a beautiful sight, and frankly, I’ve never been so pleased to see a blue LED light up! Now to see if the baby passes audio… It’s a tense moment as I connect the cables; but when the amplified sound of my own voice finally caresses my ears it becomes clear that even a simpleton can follow some well-written instructions and get a result! [Don’t be so hard on yourself Guy! – Ed.] Fair chance you can too. Now we can finally move onto the more important discussion: how it sounds. THAT MONO SOUND

In testing, this JLM Mono mic pre oozes quality, and it’s not just my ‘manufacturer’s bias’ speaking here either. The Mono pre offers plenty of clean gain and against my DAW interface preamps there’s simply no contest. The Mono is far more open and transparent sounding, and gives all of my mics a new lease on life. Even my humble Shure SM58 has displayed obvious sonic improvements. Switching from the Burr Brown Dual OPA2604AP op-amp to the JLM 99V was simple enough. In this configuration the Mono really shines. The 99V gives the lows more weight and definition, and the tops a lovely sheen. Joe describes the JLM 99V op-amps as “a Neve on steroids” and while I see the comparison I’d have to say it’s not quite as thick and syrupy as a Neve, but it does sound huge! The JLM Mono is a stellar pre in its own right. With its rugged build quality it would make a great mobile recording pre. Meanwhile, if the studio is your game, fit up the 99Vs and reap the rewards. If you’ve been putting off your first DIY build go grab your soldering iron and a JLM Mono Kit. It’s time for action!

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KORG MONOTRIBE News is spreading and the tribe is expanding… fast. Text: Blair Joscelyne

In 2010 Korg released a tiny pocket-sized analogue synthesiser called the Monotron [see the review in Issue 79] that has since become a cult hit. A quick search on YouTube instantly invites you into the basements of synth freaks the world over who have fallen in love with this diminutive machine. The Monotron was Korg’s first analogue synthesiser in over 25 years, featuring an inbuilt ribbon controller and simple controls. It’s capable of some wild sounds and its value as an instrument is starkly emphasised by its similarly proportioned pricetag. For less than $90 you get a synth that can howl and scream with the best of them, and an external input that uses filter circuitry based on the classic 1978 Korg MS20. With a huge cult following, it’s no wonder Korg has followed it up with its enhanced big brother – the Monotribe. NEED TO KNOW


Price $279

Essentially the Monotribe is the lovechild of the Monotron and the Korg Electribe, hence the name. With some sequencing ability inherited from the latter, the Monotribe features a monophonic synthesiser and a true analogue drum machine that offers basic Roland TR909-style kick, snare and hi-hat sounds. The drum sounds aren’t particularly amazing, and you can’t route them through the internal filters either, but it’s a nice bonus. This is all chucked into a stealthy black battery operated box that’s surprisingly sturdy and includes a speaker and eight-step sequencer. The knobs are well laid out, and feel reasonably expensive and super smooth. Without batteries, the synth weighs around 750 grams, and while not pocket sized like its little brother, it’s probably around the same size as a VHS tape (remember those!). For the younger readers, it’s a bit smaller than an iPad.

Contact Musiclink (03) 9765 6565 Pros Sounds amazing. Korg’s first analogue synth in 25 years. Filters internal as well as external sources. Inexpensive. Cons Drum machine cannot be routed through internal filters. Summary The Monotribe is a worthy addition to the Korg stable, and is part of a new movement back towards simple, fun-to-use and modestly proportioned synthesisers. A toy to some, an ‘inspired masterpiece’ to others, either way the Monotribe is destined for cult status.

Blips and beeps come courtesy of a VCO that’s switchable between Saw, Triangle and Square waves with octaves ranging from a super deep 64, up to two. There’s also a noise generator for adding analogue dirt. Unlike the Monotron, the small ribbon controller keyboard is switchable between three different modes: the ‘Wide’ setting allows you the full range of notes to create huge portamento effects, ‘Narrow’ gives you around an octave to play with, while ‘Key’ sets the ribbon controller to a chromatic scale. Much like the settings on Korg’s Kaossilator Pro, ‘Key’ mode essentially pitch corrects your performances for you, replaying the closest note to the frequency your finger has selected, keeping you in tune with other instruments – something that was incredibly difficult with the Monotron. The 12dB per octave low-pass filter is the same circuit as the venerable MS10/MS20, and the LFO is switchable

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between VCO, VCF, or a combination of both. I have lined up the Monotribe and Monotron next to my MS20 and they do sound quite different. Neither is better or worse, just different. The eight-step sequencer is used for both the drum patterns and synth programming, and by using Active Step mode you can change the length of the loop to create interesting rhythms and loop lengths. A Flux mode allows creation of synth parts that don’t necessarily snap to the eight-step sequence, giving a little bit more versatility by allowing insertion of notes in free time. JOIN THE TRIBE

As a lover of quirky noise making toys, and a regular creator of my own circuit-bent synthesisers, I had to grab a Monotribe the moment they became available. A tiny, cheap analogue synth with a drum machine and a sequencer built-in is my idea of fun! A lot of people have dismissed the Monotribe as a cheap toy designed to make silly analogue noises. Interestingly, that’s exactly why I ordered it, and anyone who thinks this thing is incapable of serious speaker-shaking analogue tones is mistaken. The Monotribe has been criticized for its inability to sync (there’s no tempo readout) but the recent release of a free iPhone app called ‘Korg SyncControl’ fixes this issue by letting you precisely sync the Monotribe to your DAW. I managed to do this perfectly with sequences in ProTools in a matter of seconds via my iPhone 4. There have also been noises coming from the big bad angry land of the internet that the Monotribe lacks a 16-step sequencer, MIDI functionality, has no CV inputs and is missing a host of other features, but for the price the complaints ring hollow. If you really need more features, go and buy a MiniMoog Voyager. And if you can’t afford that, buy Dave Smith’s Mopho, which will give you MIDI and an arpeggiator, though I doubt you’ll have as much fun. At under $300 the Monotribe is portable, sturdy, can be used to filter external sources, sounds great, and is, most importantly, fun – in a properly addictive, tweekable way. And that’s why we got into this studio business in the first place, right? For all the flashing lights and fun of creating amazing sounds? So there, I’ve said it. The Monotribe sounds amazing, and at only $300, I hereby declare it to be ‘The 303 for Gen Y’. In fact, the Monotribe is the first analogue synthesiser I’ve ever taken to a dinner party to pass around the table between the main course and desert! As my classic 30-year-old Korg MS20 looks on with its multiple thousand dollar price tag, I am here to confess that I feel a little uncomfortable at just how good the little synth sounds.



AN ARTURIA SPARK BEAT PRODUCTION CENTRE! Arturia’s Spark Creative Drum Machine tidily dovetails the power of analogue synthesis, physical modelling and samples, with the intuitive workflow of a hardware drum machine. Spark will save you time when looking for the right drum kit, rope you in immediately with its ease of use, and blow you away with its sonic possibilities. Spark includes sounds from classic drum machines including analogue emulations of the Roland TR-808 and 909, Simmons SDS-V, and Eprom-based LinnDrum, Drumtraks, and DMX machines, and more to boot. One incredibly lucky AT subscriber must win this rockingly brilliant production centre, valued at $699! For that fortunate subscriber to be you, all you need do is subscribe (or re-subscribe) to AudioTechnology magazine between 6/7/2011 and 8/8/2011, and correctly answer the multiple-choice question below:


[B] 16-track step sequencing

[C] Stick twirling

[D] MIDI control

Pay by credit card online: By Phone: call Miriam on (02) 9986 1188 or mail in the form below with a cheque or money order. Easy! The competition is a game of skill that’s open to all new subscribers (and re-subscribers) to AudioTechnology Magazine. The competition is open from 6/7/2011 and 8/8/2011, with entries judged on their creativity and humour by the AudioTechnology staff. Winners will be notified by phone and announced in the following issue of AudioTechnology magazine. The judges’ decision will be final and no further correspondence entered into. Circle the correct answer here:

[A] Analogue synthesis [B] 16-track step sequencing [C] Stick twirling [D] MIDI control

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A drum microphone kit with accessories… and no passengers. Text: Mark Woods

Drum mic kits are an interesting idea, and while there’s merit in having a set of mics specifically designed for the individual parts of a drum kit, I’ve often wondered who buys drum mics one kit at a time. Drummers in cover bands maybe? Drummers in original bands couldn’t afford them. Most sound mixers, studios, venues or hire companies already have their favourite drum mics and usually upgrade one or two mics at a time, rather than replace the whole lot – it feels too much like starting again. However, for those looking for a fresh start, or perhaps project studio operators setting up to record drums for the first time, the Heil Sound HDK drum mic series is definitely worth a look. Heil Sound makes an intriguing and distinctive range of mics and most of these are included in the HDK-8 kit. They’re intriguing in the way they defy many of the established practices in microphone design and application. Manufactured in the USA, the company only makes dynamic mics but by using very light, modern materials for the diaphragms, they sound a lot like condensers. There are directional mics with extended frequency responses and super-tight pick-up patterns but very little proximity effect. There are mics with ‘shaped’ rather than ‘flat’ frequency responses, tailored to specific applications. All of them are well made, high quality microphones sold at value-formoney prices. EIGHT IS ENOUGH

The HDK-8 is a set of eight mics complete with either separate clips or integral stand mounts, plus three clip-on tom mounts, all supplied in a medium-strength hard case. Starting from the bottom of the drums the HDK-8 kit contains one PR48 kick drum mic. This is a big, black and heavy mic featuring bold red front and side grilles. Unscrewing the front of PR48’s metal body reveals a large 11/2-inch diaphragm dynamic capsule and internal AT 82

shockmount system. At just on 7cm across, the flat face of this mic nearly fills the entire hole of many a front kick drum skin, and can be quite difficult to get right inside the drum. Thoughtfully the XLR socket has been angled towards the rear of the mic so it’s easy to connect the mic lead. The integral stand mount is chunky and attaches securely to the mic body. The tightening handle is also large and easy to use. Like many dedicated kick mics the PR48’s frequency response has been optimised for its application, with a big (10dB) boost centred around 80Hz and a strong low-mid dip from 100Hz to 500Hz. Reducing the low-mids on kick drum mics has been a standard EQ trick used in live sound for many years and creates a fatter sound. In the ’70s and ’80s this reduction was usually at around 300Hz – 500Hz, but in the last couple of decades fashion seems to have moved it lower, 150Hz – 300Hz, depending on the drum and musical style. Clicky kicks are more popular now too, and for this it’s common to add a second mic, often a condenser, to get a good attack. For this purpose, the PR48 also features a boosted 2kHz – 5kHz range before the response falls away quite sharply in the very high frequencies. It can, of course, be EQ’d further to taste but in use the PR48 sounds pre-EQ’d and delivers a big, deep thump down low with a good fast crack in the high-mids that may eliminate the need for a second mic. Like all Heil mics it also offers excellent off-axis rejection and no sign of overload. Stated maximum level is 148dB. TWO LITTLE MICS

Moving to the centre of the drums the HDK-8 kit provides two Heil PR22s with the suggestion they be used on the snare and hi-hat. This is an example of how Heil mics are different. Using the same mic on snare and hats is unusual for starters, moreover you’d normally use condensers if you did, but the PR22 is an interesting mic. It’s essentially

They’re different to other mics and take some getting used to, but after a while I found I started to hear their little voices saying “try me, try me”

a hand-held dynamic vocal mic with a black steel body and grille with the usual foam insert to reduce plosives. It’s also got a separate clip so the mic can be easily removed from its stand for hand-held use. Its frequency response is wide at 50Hz – 18kHz with little proximity effect up close so it’s deep but not boomy. The mids are somewhat hollowed out with a low point around 700Hz – 800Hz rising to an upper plateau around 4kHz – 8kHz. This peak in the high-mids occurs at a frequency range higher than most dynamic vocal mics and really brings out the crack of the snare. On hi-hats the PR22 is detailed and bright with no audible distortion. Its extreme off-axis rejection (claimed to be up to 40dB) helps minimise spill so both the snare and hi-hats sound as if they’re the only thing in their respective spaces. An even better application was to use one PR22 above the snare and another underneath. I needed another mic on the hi-hat consequently, but this double-miked snare sound was tight, powerful and bright without needing any EQ. As a vocal mic the PR22 is very bright with a flat, deep low-end, good gain before feedback in the monitors, and effectively controlled plosives. It makes an SM58 sound boomy and dull but it was hard sounding on some vocals, and sibilant as well. For backing vocals, however, it was a winner with its clear articulation and tight pattern making it easy to hear the BVs without having to go looking for them with the fader.


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The PR28 is designed for drums, especially toms, snare and bongos in all their various forms. The mic uses a dual suspension system to reduce unwanted noise when it’s hard-mounted to the toms and transmitted through the stand. It features a high output level, extreme level handling (148dB SPL) and the expected tight cardioid pattern. Its 55Hz – 18kHz frequency response is close to flat from 100Hz to 1kHz, rising to a peak around 3kHz – 4kHz then falling rapidly above 8kHz. In use these mics are accurate and detailed with an ability to sound just right flat (no EQ) on rack and floor toms. I used them on congas when mixing Ganga Giri one night and appreciated how they isolated the drums from the loud stage sound and provided a natural colour that complemented the heavily electronic band sound. The PR28 also sounds good to talk into and although it might be a little dull, and too prone to popping for vocal use, it would work well on brass or guitar amps.

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Moving up to the toms the HDK-8 package provides three Heil PR28s for this purpose. These look like mini versions of the PR48 kick drum mic with the same red grille inset into the flat front and round sides of the steel body. Included in the kit are three HH-1 tom mounts that are height-adjustable and connect easily to either the rim of the toms or the mounting hardware. Tom mounts save mic stands but I find they can change the sound of the drum, especially the heavy ones, and I usually prefer to use separate stands. The Heil HH-1 mounts and PR28s, however, are both quite light so any tone changes should be minimal.



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Covering cymbal duties in the collection are two PR30Bs. If I had to choose, I’d say these are my favourite Heil mics. They are an unusual physical design in that they look for all the world like side-address, large diaphragm condensers. In actual fact they’re an end-fire dynamic mic that Heil has thoughtfully attached a little symbol to – an arrow featuring

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Twin kick drums aside, the Heil Sound PR22 can cop a battering (from above or below).

NEED TO KNOW Price $1999 Contact National Audio Systems (03) 9761 5577 Pros Distinctive sounds. Excellent off-axis rejection. Minimal proximity effect. High SPL handling. Many uses beyond drums. Good value. Cons High handling noise. Summary This eight-piece all-dynamic Heil drum mic collection offers great tailored responses suitable for specific drums, co-ordinated looks, tom mounts and a hard case. The mics have great feedback rejection, tight responses that are relatively deaf to their surroundings and versatility beyond the drum kit.

the words: “end fire element” around the back of the top rim of the head of the mic, making it clear which direction to point it in. And because it’s a Heil mic, if you try to address it from the side you won’t hear much at all. The front of the grille is red, to match the PR48 and PR28, while the rest of the body is black. The stand-mount is a beauty, the best of the range. The round shaft on the bottom of the PR30B slips into a tightening ring that grips easily and securely. The PR30B uses a 11/2-inch diaphragm, and again, a Sorbothane shockmount that separates the capsule from the body of the mic. The frequency response is wide at 40Hz – 18kHz and quite flat from 50Hz – 10kHz, with a 3dB presence peak centred just above 4kHz. The pick-up pattern is closer to hyper-cardioid with particularly strong rejection from the sides. This tight pattern is partly why these mics tend to work so well as cymbal mics; the separation of the cymbals from the rest of the kit is pronounced compared to the usual small diaphragm condensers typically used as overheads. The transient response and fine detail are comparable to a small diaphragm condenser and I liked the way the cymbals could be hit hard without turning harsh and nasty. Also impressive was the way it could pick up the low-end of cymbals, particularly the ride cymbal. But my favourite use for the PR30B was not on drums or cymbals. Everyone is looking for that ‘big, fat’ sound from guitar cabs and here this mic delivers in spades. Clear mids, strong low-mids and accurate high-mids – guts with clarity but a minimum of ear-bleed. It worked on every amp I tried including Tim Rogers’ full and dynamic playing style, Ash Grunewald’s louder Fender Twin amp setup and Lucie Thorne’s super-warm amp sounds. But wait, there’s more! The PR30 is also a very good vocal mic. Not for handheld live use, it’s too bulky for that, and

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it does have high handling noise, but it’s big, rich tone and clear articulation would be great for broadcasting or announcing applications. GREAT PR

The more I got to know these mics the more I liked them and everything I’ve said about their live qualities applies to recording too – except the part about gain before feedback! I rarely use dynamic mics at all for recording but after using the Heil mics at several live shows I was confident enough to record a whole African percussion album for Sayon Souare using only mics from the HDK-8 kit. The PR48 got the big drums, the PR28 was used on the mid-sized skin drums with the PR30B on kri, shakers and lead djembe, while the PR22 worked well on the fairly sparse vocals. Many of the instruments were so loud they made my head snap back if I walked into the recording room while he was playing, but the Heil mics have an ability to tame loud peaks and present the recorded sounds in a close, warm way with no distortion and almost no room sound. They’re different to other mics and take some getting used to, but after a while I found I started to hear their little voices saying “try me, try me” on a variety of sources including bass cabs, fiddle and accordion. Once you get to know them it’s unlikely the mics in the HDK-8 kit will be restricted to drum duties. Maybe the PR48 kick drum mic will be, but the others are all interesting and distinct sounding general-purpose mics that will also do a nice job as problem-solvers, particularly in situations where good separation is required. Compared to the relative cost of what most people spend building up a set of decent drum mics, the retail price of the HDK-8 kit is quite a bargain. Also available is the HDK-7 kit, which is much the same but has one less PR28 tom mic and mount. Check ’em out.


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ELECTRO-VOICE RE320 The voiceover king is now the king of kick. Text: Al Craig

NEED TO KNOW Price $399 Contact Bosch Communications (02) 9683 4752 Pros Great on instrument or voice. Rugged construction. Two distinct frequency responses. Three-year warranty. Cons I really can’t think of any negatives with this mic. Summary If you suspect you need a quality dynamic mic in your cupboard, the RE320 would be a very good choice. It’s legendary for voiceover work, works particularly well on kick drums thanks to its tonal shaping and directionality, and is versatile on a wide range of instruments.

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My first real experience with the Electro-Voice RE20 was way back in the late ’70s when I toured with Blood Sweat & Tears and used it on kick drum. Back then it was the mic of choice for bass drum – it was always on my production rider – and these days I own two of them. They’re still in production today, of course, and EV must have sold countless thousands of them over the decades. Now the 320 has arrived, building on its parent’s reputation as a genuine classic. The original RE20 is a bit of an icon as far as dynamic mics are concerned. In particular, the mic has become synonymous with voiceover work and radio DJ’ing, so much so that it’s prominent on at least two TV shows that include a radio station in the story line. There’s a good reason for this, of course: it may well be the most commonly used announcer mic in radio stations the world over – it’s looks, sound and shape are now literally inseparable from the role it seems! Here in Australia the RE20 sits pride of place at 2CH, MMM, 2DayFM and countless others. AN ICON RE-BORN

The Electro-Voice RE20 and RE320 are the biggest largediaphragm dynamic mics money can buy, the RE320 being the newest generation of voiceover classic made by the longstanding US manufacturer. It’s a little lighter, shorter and brighter than the RE20, and owing to its neodymium magnet, puts out a little more signal (a whopping 1mV/ Pascal). It’s also black nowadays and comes with a switch on the body that tailors the frequency response of the microphone specifically for kick drums by giving it a considerable (if you’ll pardon the pun) kick in the bottom-

end response. For so many years the RE20 has been used to mic up kick drums even though I’m pretty sure EV never anticipated its use in this role – now the new Chinese manufactured RE320 caters to the bass drum specifically. Essentially flat from 100Hz to 4kHz, with a little dip at 4.2k rising +4dB at 5k, dipping down again at 7.2k and then rising again at 10k, the unadjusted frequency response of the RE320 is primarily tailored for voice pickup. When the ‘kick switch’ is engaged a different response curve is applied, which sees a rise at 125Hz, a dip of 4.5dB at 380Hz (à la your average engineer’s kick drum EQ low-end shaping) with additional peaks at 2.8, 4.2 and 7.2kHz. This ‘kick switch’ effectively gives you two mics for the price of one (although it can’t be in two places at once for obvious reasons). The polar response is also very even up to around 8k, which is very handy when you have less than ideal mic technique. THE VARIABLE-D

Electro-Voice secured patents for ‘Variable-D’ technology way back in 1963 when it first introduced a range of mics featuring the technology and trademarked the name. Other mics in the range back then also included the RE664, the RE16 (hyper-cardioid dynamic), the RE15 (cardioid dynamic) and a couple of cheaper cousins (the RE10 and RE11). But the flagship model that has sustained throughout the years was the RE20, followed on years later by the RE27ND. So what is ‘Variable-D’ exactly? Basically, by careful design of the rear entry ports on the body of the mic, EV was able to greatly reduce proximity effect common to directional

PROXIMITY EFFECT: FAULT OR FEATURE? What happens with most directional mics is that there’s a natural tendency for the sound to go around the mic and in the ‘back door’ ports that are there to aid control of the polar pattern – especially lower frequencies. The closer the source is to the mic, the more prominent this phenomenon becomes. This is neither a good nor bad thing per se, although, technically speaking, this proximity effect distorts the source with added bass. Omni mics don’t react like this of course, and in the end, the phenomenon just becomes another tool in an engineer’s kit-bag. A good sound engineer will take these (and other) factors into account when selecting a mic for a specific application. Some singers

insist on having a mic that offers proximity effect as they can use it to form part of their sound. Proximityprone mics can be used to draw out the warmth on a thin guitar, fatten up bass amps and snares, even make a smaller kick drum sound bigger. Horn players dig them because they can use them just like the singer and if they need that little extra punch in the guts they can lean into the mic and bellow.

But broadcast audio engineers need a directional mic that has the least amount of proximity effect possible so that radio station DJs sound natural no matter where they work the mic. Voiceover artists generally have a natural beauty in their low-end registers anyway that doesn’t require colouration.

microphones and provide a balanced and uniform frequency response up to 180° off-axis. This impressive capacity not only made the RE20 great for voiceover work, where DJs tended to move around a bit, in a rock ‘n’ roll setting where vocals were recorded live in a room full of loud instruments, the resulting off-axis spill into the vocal mic sounded natural, not horrible. RE20 VS RE320

Now, onto to the inevitable comparison between these two mics. Even after all these years, I still regularly break out my RE20 to do corporate voiceover work. One of the main reasons for this is that my voiceover booth has a nasty standing wave in it at about 422Hz, and certain voices and vowels seem to exacerbate this annoying resonance. Consequently, I really need a voice to work the mic quite closely, and this is where the RE20 fits this bill perfectly thanks to its immunity to proximity effect. If I use a directional mic with proximity effect in the booth, I get an unnatural buildup of the low end, which obviously doesn’t suit the mission at hand. I also often use the RE20 for tracking individual backing vocals for much the same reason. I put up my trusty RE20 (which was re-conditioned by EV two years ago) and the new RE320 side by side and recorded several different sources. In all cases, I found the signal from the RE320 to be about 5dB hotter (as expected). On a simple voiceover, after adjusting for the gain differential, I listened to both mics in turn. The RE320 was much warmer and very pleasant to listen to. There was very little colouration at 90º off axis and it also seemed to handle transients a little better. The ‘presence peaks’ worked a treat too; offering more clarity than the RE20, while still retaining that signature ‘RE’ balance. I also recorded a band using the 320 on the kick drum (surprise surprise) with the new ‘kick switch’ engaged. Wow! Normally, I’d be reaching for the input EQ during a tracking session and pulling and pushing frequencies here and there. With the RE320, however, it started out flat and stayed there. It wasn’t until I got to mixdown that I need to make some tonal adjustments, and only then to make the kick and the bass fit together. CONCLUSION

Microphones are very personal objects of audio desire, and like shoes, you can never have enough of them. Any working mic will deliver a unique sound that may (or may not) enhance a project, and unless you have several to choose from, you’ll never know that ‘Microphone X’ was perfect for Johnny’s tuba solo even though it cost 20 bucks and was deemed good for nothing. Choosing the right mic, placing it in a good space and having a great musician is the secret to delivering great results. Having said all that, if I only had two mics in my cupboard, the RE320 would be one of them.

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WAVEMACHINE LABS DRUMAGOG 5 ‘PLATINUM’ When your drum tracks suck the big kahuna, it’s time to give them a damn good ‘gogging’. Text: Calum Orr

Eight years ago I discovered a plug-in that has since saved many an ill-recorded drum session from the virtual dustbin. Over the years, sessions sent to me to mix by other engineers, or sessions I myself have recorded, have regularly been given ‘the Gog’ drum replacement treatment (as it’s come to be known at my studio), whether it be to augment the drum sounds, replace them altogether or a mixture of both. Quite often my clients have become so enamoured with the program that they’ve later quipped: “Hey Cal, can you ‘Gog’ the drums on the next song too please?” NEED TO KNOW Price US$379 Contact Drumagog Online Pros Improved GUI. Advanced features. In-built effects. Great new filter section. Cons Still can’t import standard file types. Summary If you mix records for a living, or even for your own amusement, but haven’t tried Drumagog, check it out. The plug-in is capable of utterly transforming poorly recorded, flat and lifeless drum sounds, and salvaging mixes that may have otherwise seemed destined for the scrap heap. If ‘salvaging’ is a word that aptly defines your approach to drum mixing more often than not, you’ll love Drumagog 5.

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Version 4 of Wavemachine Labs Drumagog is hands down my most oft used plug-in. While the program has saved many a mix from mediocrity, the hit detection algorithms and delay compensation facility has often meant that files have had to be shifted forward in time by small amounts of samples to maintain phase coherence (a small price to pay for the vastly improved tones it provided). Armed with this knowledge and myriad requests from a huge user group, Rim ‘Shot’ Buntinas – Wavemachine Labs’ main man – has since addressed these issues, and v5 was recently released to wide acclaim. Once purchased from Wavemachine Labs’ online store, the Drumagog download itself is 127MB, while the proprietary ‘gog’ files containing thousands of multi-layered drum samples is around 1.2GB. Installation on Mac systems comes in AU, VST and RTAS flavours, and on Windows systems, VST and RTAS. Authorisation is via a Pace iLok key (not supplied). NEW FEATURES

At first glance, the new Drumagog GUI is impressive, and has a decidedly ‘pro’ look and feel about it. The new facelift includes many new features including an advanced triggering engine that can distinguish between left- and right-hand hits, and even open, closed or semi-open hihats! Just like in v4, getting the replacement happening is as easy as moving the sensitivity slider up and down to ‘catch’ the required hits. There are also Transient Detail and Resolution sliders to enable more precise viewing, as well as input and output faders for the source sound being triggered and the replaced output file. So as to eliminate other sounds – like bleed from other drums triggering the samples – the program also now features a comprehensive

filter section for the trigger, that includes high- and lowpass, band and band-reject filters that have adjustable ‘Q’ controls for really honing in on the source sound. The filters really come into their own when the tempo is quick and the drum recording contains lots of flourishes. Another great addition in v5 is the ‘Random Multisamples’ option, which – as the name suggests – randomises the drum samples being replayed. To my ears this option produces a far more realistic and natural result. The function runs deep too, with the user having the ability to set groups of samples to trigger at varying velocities. On the other hand, if you want to have greater consistency across a track, you can apply the ‘Dynamic Tracking’ tool, which makes the samples more regular and smoothes out the performance. This feature works particularly well in songs that have a more straight-ahead rock feel. A new effects section, featuring a built-in convolution reverb, adds yet more strings to Drumagog’s bow. This section allows you to create realistic spaces for the samples without having to leave the plug-in, and sports 30 impulse responses to get you started. It also gives you the option of importing your own IRs later. The same effects section also incorporates the new ‘Morph Engine’, which is great for adding odd flavours to the mix in, say, a breakdown or an outro. I haven’t explored this feature’s potential with any great gusto yet, but I’m sure the Morph Engine will be put to good use by the bold and the electro-leaning. Likewise, the drum synth section lets you make alternate versions of your drum sounds, which will no doubt prove invaluable for re-mix engineers, dance producers and the more adventurous mix engineers among us. Saving the best to last, probably the most ambitious new feature in Drumagog 5 is its ability to trigger other drum software, like FXpansion’s BFD, Toontracks’ Superior Drummer and others, from within Drumagog. This is a fantastic timesaving feature, vastly more practical than the original workaround measure, which involved converting audio to MIDI, which was all a bit mind-numbing for yours truly. There are many other features – like auto hi-hat tracking, greatly improved preset management and a fantastic new interface – that make Drumagog 5 a no-brainer upgrade for existing users and a must try for those who haven’t ‘gogged’ yet.

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Why wait around for a flash of inspiration to strike when the simplest way to fan the fickle flame of creation is to just use your hands. Text: Andrew Bencina


I find recording music a pretty good way to avoid facing the reality that I’m getting older. If you live and work within a time capsule full of vintage equipment that was first tagged and tested before you could talk, it’s not so hard to feel young. Unfortunately, the edifice of this fantasy is beginning to crumble as the cultural cycle wheels around. If everything old was new again I could probably maintain the facade, but just lately many things I associated with my eternal youth have returned to fashion, and in a cruel instant I’m 20 years older. So it was when I found out I’d be reviewing a new drum machine from virtual instrument virtuoso, Arturia. Quite honestly, I had to pack up my Korg DDD-1 (circa 1986) to make room. The Spark Creative Drum Machine – as it’s been confidently christened – is the very latest in a line of hybrid software/ hardware instruments from the popular French synth modeller. Spark may be one of only a few current hybrids to fully embrace such a high degree of integration (Native Instruments’ Maschine springs to mind as another) but there seems to be little doubt about the potential of the development model. Instead of filling your studio, or roadcase, with countless gizmos, each requiring its own power supply and audio interconnect, why not make the most of the latent power within your expensive computer, avoid unnecessary stages of digital conversion and get hands-on with physical interfaces crafted to unleash your creativity. AT 90

In the most basic of terms, Spark is a 16-track stepsequencer for both Windows and Mac paired with a percussion sound library and custom hardware controller. The software is provided as both a standalone application and plug-in instrument, compatible with your choice of host (VST; 32 & 64-bit, AU, RTAS). The sound catalogue is divided into three categories: Analog, Physical Model and Samples, arranged in a series of factory kits. Synthesis is handled by Arturia’s ‘True Analog Emulation’ engine. As a result you’ll find lovingly-crafted recreations of many of the classic analogue beat boxes here – 808, 909, 606, CR-68 (Vintage Box) and Simmons (Sci-mons). I wouldn’t say they all precisely capture the sonic-patina of 30-year-old electronics but the sounds nevertheless deliver a satisfying modern representation of each, with twiddle-room to personalise. The analogue emulations are complemented by selections from third-party developers like Ultimate Sound Bank, Sonic Reality, Ueberschall and Modern Beats. These include classic drum machines from Sequential Circuits, Oberheim, Roland and LinnDrum, as well as a range of acoustic and electronic kits. It’s also possible to load your own samples and create custom kits of any combination of instruments. GRID SEQUENCER

The pattern sequencer, located in the top pane of the software interface, is of a standard gridded step design. Patterns of up to four bars in length can be sequenced with a step resolution of up to 1/64th of a beat. Each instrument features a selection of editable parameters particular to it, and these can be sequenced via automation channels within the pattern editor. Mixer parameters like Volume and Pan can also be automated here on a per-pattern basis. Spark ‘Projects’ accommodate four banks of 16 patterns which can each be combined within a larger ‘Song’ sequence via

an intuitive drag ‘n’ drop interface. Songs have a maximum length of 64 patterns and it’s currently only possible to store a single song per project. In the lower pane of the interface you’ll find a per-project mixer (featuring a creative selection of insert and send effects), a studio window for selecting and customising kits, and an iPod-style scrolling Library for loading projects. PUSH THE BUTTONS

While the bulk of Spark’s power resides firmly within the computer it’s the Spark controller that will have you salivating. The control surface boasts approximately 30% more surface area than my Korg ‘padKontrol’, and weighing in at nearly 3kg it’s also a damn sight heavier. However, with mass comes solidity and a rugged plastic case has been elegantly complemented by a thick metal top panel. The base of the unit is uneven – featuring a hump-backed pod to house its connectors – while its body rests at an ergonomic incline astride two sleigh-like blue plastic runners. For me these slightly flexible legs are the chassis’s weak point and make Spark seem far better suited to a permanent table-top installation than being passed from lap to lap on your studio couch. Spark, along with Arturia’s ever expanding range of synths and controllers, is dressed in sparkling white and is overflowing with backlit pads, buttons and continuous rotary controllers. The rear of the unit features its primary USB interface connection, MIDI I/O (the controller can operate as a MIDI port for other applications), and an optional – and largely surplus – 9V DC power socket. In addition to the main Spark software application, a MIDI control centre programs the hardware for control of thirdparty software and devices. This MIDI editor provides an impressive level of configuration, albeit lacking in some other facets of the software. For instance, Arturia has notably neglected to include a selection of velocity curves for pad response (also missing in Spark itself) but delivers in most other areas. The controller can switch between its Spark and MIDI modes at the press of a button (well three buttons simultaneously to be precise) and supports configuration of 32 trigger pads, 78 rotary encoders, 48 buttons, and three X/Y pads in any one MIDI template. While the Spark controller is unquestionably an impressive and flexible MIDI device it’s worth noting that the software’s MIDI generation capabilities are currently limited to MIDI export of pre-programmed sequences. PLAYING WITH FIRE

The best way to experience Spark is to press Play on the controller and improvise. The pads respond consistently across their surface and the visual feedback provided by their synchronised illumination significantly enhances live sequencing. Recording and then erasing performances without interrupting playback is seamlessly handled, and the inclusion of a row of 16-step buttons for detailed editing of individual tracks is most welcome. Dedicated controls for live slicing, looping and filtering effects only increases the potential for fresh moments of inspiration – although these cannot be recorded for posterity unfortunately. I actually took the time to configure a few other third-party pad controllers to run the Spark software and none of them went even close to equalling the experience of using the tailored interface. Saying all this, it’s important to note that as a new product –

I was testing the only unit in the country at the time – Spark does have its fair share of teething problems. In fact, if I were to focus on issues of user configuration and deeper hardware control of the software I would quickly run out of space here. The one advantage of the hybrid model, however, is that these issues can be rectified relatively easily via software updates. However, the hardware controller has unfortunately been lumbered with a few design limitations; namely a retro-looking 32-character display that’s difficult to read when scrolling, and a lack of generic modifier buttons or rotary controllers incorporating push button switches. These more permanent omissions don’t necessarily preclude future improvements to integration but they certainly complicate them. I could go on, but as Arturia is providing an almost fully functional demo version of the software – only saving and exporting has been disabled – from the updates page of its website I’d recommend you try it for yourself. Rest assured, if you like what you hear, the experience will only be further enhanced by the use of the controller. MONEY TO BURN

When you consider that Spark is barely one quarter the price of the current cream of hardware drum machines – not to mention some vintage options – it’s inevitable that its Gallic charm will prove irresistible to a whole host of beat-niks. Though it may be a work in progress – intentionally so in some respects – the hybrid model adequately accommodates this future development and offers a sense of longevity and security for your investment. If Arturia can bypass the superficial restrictions imposed by the controller’s labelling and trade in its left-bank cafés for some left-field thinking there’s no telling how much further the software/hardware integration could be pushed. Spark is an exciting new player in the performance sequencer market, but only time will tell if it reaches the classic heights of those it currently emulates and no doubt aspires to replace.

NEED TO KNOW Price $699 Contact CMI (03) 9315 2244 Pros Intuitive controller interface encourages expressive performance. Diverse tweak-able emulations, physical models and samples. Software core enables ‘infinite’ future development. Cons Too many software features unavailable via hardware controller. Sampled Instruments under-developed by current standards. Controller design may limit product evolution. Summary With Spark, Arturia has delivered a tactile and interactive drum machine with a largely unrestricted potential for future software development. As a young product, this initial release leaves plenty of room for refinement but anyone with a love of classic rhythm boxes of yore will find an immediate connection to this fresh yet familiar instrument.

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(…doing alright ’cause we’re making money) Text: Rick O’Neil

I’ve lived in the same town of Sydney my whole life. Actually, when I say ‘the same town’ I mean the same country and even that’s not strictly true. And when I say ‘my whole life’ that’s not strictly true either. I’ve lived in other parts of the world for brief periods but am most certainly a Sydneysider when all is said and done. In recent years I’ve been fortunate enough to travel far and wide thanks mainly to my wife, who has dragged my reluctant-to-leave-thestudio-for-an-unpaid-holiday arse literally to the four corners of the globe, to some of the most surreal places on earth. If you ever want to know what Antarctica’s like, for instance, give me a call. But put aside the rest of the day; it’s some tale. Recently we went to Greece on one of those trips that kind of resets your thinking. Spending time in bankrupt Athens reminded me of that song about the Lucky Country. It’s true, we don’t know how lucky we are in Australia sometimes, but when it comes to the audio industry the grass has always seemed greener on the other side. THE LUCKY COUNTRY

I still remember lying awake at night years ago (back in the day when I was one of the lucky few who had a full-time job in a recording studio), wondering about the great studios of the world, imagining walking into Abbey Road, Ocean Way, A&M, Capitol or The Townhouse for the first time. My job mastering at Festival meant tapes would arrive in my cutting room from these mystical studios on a regular basis. As I laced these tapes onto the mastering machine, and with their multi-coloured boxes in hand, I would always read the notes to check out which rooms they’d been recorded in, who the producer was, and muse over the fact the last time I got a tape from XYZ studios, Johnno was the assistant who wrote his name in the bottom corner. Now he AT 92

was the engineer and in a year or two he would probably be the producer. I dreamed of ditching my regular studio income, getting on a plane and mixing it with the big boys in the glamorous world of freelance engineering and production. I wanted my name on those coloured boxes from overseas – imagine being able to charge a dollar a minute rather than earning $140 a week! Wowee… the sky was the limit. From time to time somebody you knew would make the leap; jump on a plane to go and work in these mystical places – maybe a band you knew or local sound guy – and occasionally international guys would come Down Under to record, and tell you about all the magical things that lay across the ocean. The grass was very green over there in the United States and England. FULL-TIME SYNC

Once a year my full-time commercial studio job would allow me a few weeks off with pay, usually over Christmas, and every year I would drive up to Bryon Bay and act like I was 19 or 20 again, even though I actually was 19 or 20 back then. At work you had to act 35. I’ve been old before my time most of my life. Anyway, when Christmas holidays were over I’d check my bank account and discover that the grand old stoic record company had paid my first week’s wage of the year in advance, which gave me just enough money to drive home again! I had almost no appreciation back then of what a luxury that regular income was – minimum wage every week, year in, year out! Earning big money as a freelance engineer seemed like a more attractive proposition at the time, of course, but these days the feast or famine oscillation of that income stream can be a little hard to take.

By the time I did finally make the break from the regular studio job I experienced the timehonoured muso tradition of trying to go on the dole, so I could afford to eat while I got my ‘career’ together. But getting on the dole was too hard – I hadn’t realised you were supposed to tell lies to get your payment. So I scratched around doing other music related jobs: working at a guitar store, doing live sound for bands, recording one of these bands in a bomb-shelter studio for lunch money. I even did a couple of lectures at an audio school at one point! I guess many of you reading this already know the truth behind the ‘glamourous’ life of the freelance engineer – sometimes you get to the point where $100 is the difference between eating well for the week or not eating at all, so you set your sights low that week – real low. At this point in your career the thought of jumping on a plane to go visit Studioland International quickly becomes a non-reality. When a plane ticket costs $1800 and you’re not making that much money in two months, minus rent and food, there isn’t really a choice. Even just keeping the car registered stretches the imagination – you just do what you can for whomever you can. CASH NETS

The thing this period of my life taught me was that nobody wants to pay a guy that will work for free, and everybody wants to work with the guy they can’t afford to pay. But we all want to live the rock ’n’ roll dream no matter what it costs, and thanks to my father who ironically worked in banks, I learnt the value of a dollar. Eventually I found a way to lay out some ‘Cash Nets’ and haul in some clients so I could get remunerated for the work I did, rather than pay for the privilege of recording them.

My dad will laugh if he reads this, seeing me preach about ‘earnings’ like this, but there’s no getting around it: to make enough money to get by you have to lay out some Cash Nets and check them regularly. Then once you’re established you have to make sure you tend to them constantly, and never grow complacent of the income they provide. The main problem with freelancing is that the work sometimes keeps you from maintaining your nets properly and before you know it they’re full of holes. If your head’s forever buried in the console there’s usually precious little time to answer phone calls from prospective clients who will be paying you two months from now. Then one day you wake up to find that the phone’s stopped ringing, the money from your last job has hit your bank account and left again the same day – that’s when you know things are getting tight – and the clients that couldn’t get through to you last month have gone elsewhere. I tend to joke about it but I suppose I’ve never been truly broke, but boy have I been more than a little bent out of shape a couple of times – probably will be again. CASH NET NO. 1

So let’s talk about a sound person’s Cash Net: where to lay them, and more importantly, how to maintain them over the long haul. The first and most important thing to establish as quickly as you can is what you’re really worth. Not what you imagine, or dream, or scheme up, but what you’re worth. The clients who you expect to earn money from won’t be factoring in how much cash you shell out in rent, how much credit card debt you’re buried under or how much your car costs to run. They’re too worried about their own circumstances to care too much about yours. This is the biggest mistake I’ve made over the years: I would often set my rates based on what I thought I needed to get me through the week – it’s short term thinking and it’s dumb. Truth be told, I probably still do it – but there’s nothing like telling your neighbour how to mow his lawn when yours is six feet high is there? The trick is to find out how much somebody whom you’ve never met is willing to pay for your services – no mates’ rates, no sweetheart deals, just how much your services are worth… at a minimum. Work out that figure and write it down, and consider it one of the most important lessons you can learn. How do you find out what the minimum is you should be charging? Well, let’s keep it simple: the minimum adult wage in Australian is $570 a week, or about $15 an hour. If this figure comes as a shock to you, just remember that close to half the Australian population works for that sort of money, week in week out. If it sounds like a luxury to you, read on; you may be in need of some help. Most jobs in Australia require you to simply show up and work. There are no expensive tools required and you get $115 a day – and in the modern world ‘a day’ is defined as 7.5 hours. So guess what, that’s the minimum you should charge for your audio services, at the very least.

Let’s not forget that your ‘standard day’ is almost certainly longer than the average – sometimes twice as long. And let’s not forget how much money you’ve invested in all those tools you’ve bought just to get the job done. Surely they add value to your hourly rate, right? In truth, you might be able to charge way more than the minimum wage, but we’ll get to that in a minute. Meanwhile, if you’re working for less than $15 an hour, you’re certainly not the first person to work for so little in the audio industry, but it’s not enough. I reckon the Australia minimum wage is a good bottom line none of us should cross. Okay, so now you know how little you should charge – and forgetting for a moment how much more you might charge – let’s have a think about where your money is going to come from. Do your clients have any real budget restrictions or are they just trying to pay you the minimum they can? In my experience clients are always trying to pay the minimum for all their audio services, so the next lesson to learn is how to get as much of your client’s money as you can – fairly of course – as often as you can, while still maintaining the highest professional standards you can. This is a very important point to understand: sure you can play on their record, mix, master and probably produce your clients’ record. You might even be able to do the artwork, distribute and promote it for all I know, but are you really that good at all those things in equal measure? Let’s call a spade a spade here. You’re not. If you try and do everything for your clients, and then suddenly they decide there’s something wrong with one aspect of your work, chances are next time they will go elsewhere for everything. Moral of this story: understand what you do well, and make that your primary Cash Net. We’ll talk more about diversity in a minute but the key to surviving in this business long term is to get better and better at what you already do best. The age-old mistake is to try and be everything to everybody. I can’t tell you the number of sound people that have said to me: “Oh, I don’t do real mastering, I just do it for the bands that don’t have a budget for it. I always recommend they go to a ‘proper place’ if they can afford it.” I could write 10,000 words about that here if I let myself, so let’s move on. In terms of your primary Cash Net – the one you specialise in – work out the minimum amount of money you can charge, and try to identify who your clients are and what it is they need from you. Remember, it’s not their problem how you pay your bills, so the obvious thing here is to charge what they are willing to pay and get as many of them to pay this amount as often as you can. Now let’s do the maths: THE SUMMING BUS

Multiply the minimum you can charge by how many times you realistically think you can charge this amount per month. This will give you a

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baseline (not a bass line) financial projection of what your primary Cash Net hauls in. It’s amazing how many freelance engineers there are out there who have never thought to do this. Once you’ve made this calculation, if the figure seems low you’ll understand why that credit card debt is mounting up, and perhaps start thinking about what to do about it. Now, of course, if you’re in demand you can charge more than the minimum – way more in fact – but there’s a point you can reach where you suddenly price yourself out of the market. This is a dangerous limit to exceed, because once you do, your Cash Net will start to tear and times will get tough. Now this is not supposed to be a lesson in economics, trust me, I’m the last person you want to learn economics from, but as a lesson in survival in the music business, I reckon I speak from a pretty solid footing. CASH NET NO. 2

So, let’s say your primary Cash Net isn’t quite working out for you – now is the time to diversify, and find the other ways to earn money. You’re reading one of the ways I do it right here, although if I really did this for money I’d have stopped these rantings long ago. My primary Cash Net is certainly not as a pro audio magazine writer, it’s just something I do to keep my nets tended. There’s certainly some branding advantages behind putting yourself out in the public eye like I do, but obviously masquerading as a journalist isn’t for everybody. Lots of sound guys I know keep themselves in work by playing in bands as a sideline. Quite often those bands (or friends of those bands) send them work, and $50 petrol and pizza money two nights a week is better than none at all I reckon. If you’re a live sound guy you might set up PAs in hotel conference rooms. Whatever work you can muster in your related field works well because it keeps your profile up in that community, and feeds back into your primary Cash Net. Jobs pop up all the time through this related work. The other option of course is to get a good old-fashioned non-audiorelated part-time job. The point here is that almost all the ‘lifers’ in this audio game have moonlighted at some point or other, and those that haven’t probably should have! CASH NET NO. 3

So, as well as your primary Cash Net you might have anywhere from one to five Diversity Nets. The thing all these nets have in common is that they earn you money. It doesn’t need to be a lot and they don’t need to be consistent either. They just need to be real. I can’t tell you how many times one of these Diversity Nets has landed me unexpected income seemingly out of the clear blue sky – right when I needed it most. Of course, if you don’t expect to get money for your services, primary or otherwise, that’s okay – you’re an enthusiast. The audio world is full of people like you, and for many people this is the best way to be involved in the industry without burning out – turning your passion into a job. AT 94

Plenty of people don’t want it to become a job! Okay, so now that you’ve identified your primary Cash Net, you can call that your ‘job’ if you want to. The next thing to understand is that your Diversity Nets are nothing to be ashamed of – and you might as well get good at those while you’re at it. Just remember, to survive you need a steady flow of clients coming in, so you have to repeatedly ask yourself why they’re coming at all. Is it really because you drew up a gearversus-business plan? Not likely. Most likely it’s either because of your location, your skill set or your reputation – hopefully it’s a combination of all three. Your best clients will be the ones that regard you as fulfilling all three criteria – you’re their Trifecta: the right skills, the right place, and the right pedigree. But here’s the rub: as a freelancer, why would somebody hire you when you hardly have any equipment compared to the guy down the road who has lots of specialist tools. There are several reasons why people hire you of course, not the least of which are (hopefully) your people skills and amiable personality, but it’s certainly much easier to do your job if you have some decent tools, preferably ones you know backwards. Lots of freelance engineers own a bunch of specialist tools – very few have none at all. But the way the world is heading, having your own tools isn’t enough – you need your own location as well – it doesn’t need to be a studio as such nowadays – probably never did in all honesty. Lots of the world’s greatest records were made in houses, barns and churches, forsaking the traditional studio setup. But being a guy who doesn’t work for somebody else, I can tell you it’s much easier to get people in the door who will pay you money if you actually have a door for them to walk through. But that’s when things start getting more expensive for all concerned – you and your clients – and before you know it you’re a studio owner, not a freelancer. Sound familiar? Once you have a studio that requires you to charge more than the minimum wage to keep the doors open, a strange thing happens: you spend more and more time working out ways to keep people coming through the door, and less and less time going out the door yourself. Freelancing becomes a thing of the past and suddenly you’re a studio owner. Visions of flying around the world are replaced with the reality of all the bills a studio generates, and as I mentioned earlier, if it weren’t high on my wife’s list of priorities to make sure we travel overseas once in a while, I’d probably never leave the studio. I’m always watching the cashflow you see, and an unpaid overseas trip always feels like the wrong type of flow – cash out the door! Smart woman my wife. There’s a rare breed of freelancer that owns no audio gear, stays in work year in, year out and never has to worry about their Cash Net. But they’re atypical. The rest of us meanwhile have to keep on keepin’ on. Rick O’Neil runs Turtlerock in Leichhardt, Sydney… where the door is always open.

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