AudioTechnology App Issue 17

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REGULARS

ED SPACE HRA?: Wot Evs Gramps Column: Christopher Holder

Editor Mark Davie mark@audiotechnology.com.au Publisher Philip Spencer philip@alchemedia.com.au Editorial Director Christopher Holder chris@audiotechnology.com.au Art Director Dominic Carey dominic@alchemedia.com.au Graphic Designer Daniel Howard daniel@alchemedia.com.au Advertising Philip Spencer philip@alchemedia.com.au

Editor, Mark Davie, has stepped out of the building to be with his wife on the occasion of the birth of their second child. Good luck Mark and your soon to be four-piece brood.

Which means I have the pleasure of stepping into the AudioTechnology Editorial chair just as I’m contemplating the apparent rise of hi-res audio (HRA).

It doesn’t take more than a minute’s musing to conclude that this is a multi-faceted debate laced with ignorance, hubris, bluster and disinformation. So I thought I’d use this page to stoke that fire with some opinion of my own.

No one has the answers, but it’s fun to explore the issues.

So let’s start with what we do know: Most people listen to shocking-quality audio: We have an infogram hanging on the wall of our editorial office that shows how a record production painstakingly documents a performance with hundreds of thousands of dollars worth of equipment, some of it lovingly maintained and repaired since the ’60s (or earlier), often for their hair-splitting, subjective qualities. And the grand irony is that those lovingly and professionally recorded sessions are enjoyed through the ubiquitous $1 Apple white earbuds. It would be laughable if it wasn’t so depressing.

The truth is: it has ever been thus. The general population has always listened to music through crap speakers or headphones. It might be one of those Gidgetstyle portable record decks for listening to the latest single in your bedroom. Or it might be listening to your Walkman via $20 Koss earpieces in the ’80s — probably no better-sounding than the current Apple buds. Which leads us into truth No. 2: Most people listen to shockingly-encoded audio: Most people listen to MP3s. If you gave them the same song in hi-res audio, could they hear the difference? Yeah, most would. But if you then told them that rather than having 1000 songs in their pocket they would only have 10, you’d soon get the brush-off.

What do we conclude from this?: People will happily trade quality for convenience. People will gladly listen to phasey/thin MP3s in exchange for having all their music with them all the time. If the audiophile loves music to the point they need to schedule their listening sessions, close their eyes and leave the rest of the world behind the sonic forcefield that is the perfect soundstage before them… then most of the population is diametrically opposed.

Instead, what the population is demanding is: ‘any song I can think of right now, and I don’t care what you have to do to make it happen’.

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So where does that leave HRA? Because this is where popular music is headed. In my view it leaves HRA in the periphery.

I’m not decrying HRA. I’ve loved the time I’ve spent listening to SACD, it can be spine-tinglingly beautiful; present; real. And especially for classical music or acoustic jazz, there are many people who would appreciate an HRA version of those recordings — as they already currently do with specialist labels such as Chesky.

But how can HRA go mainstream?

Sony thinks HRA can go mainstream, marketing a range of HRA-compatible products. The schtick is quite simple: download music in hi-def and then you can transfer to the hard drive of a Sony HRA receiver and Sony will play it back full-def, whether it’s 24/96 or DSD. Okay, cool. But this is still the preserve of the outliers… those people anal about sound.

Neil Young’s Pono has raised eyebrows as well as some start-up capital. Pono is an HRA environment: a player and a marketplace for downloading HRA music.

Pono peddles the line I’m hearing more and more: ‘music as the artist created it, the way they heard and felt it’. Already I’ve got a sinking feeling about Pono. As much as I want to champion it, I feel it has its priorities upside down. There’s a palpable spectre of an indignant Neil Young mumbling, ‘enough is a enough’, about Pono; a sense of the artist fighting back and demanding his/her art be treated with the reverence it deserves.

Bollocks. It’s not about the artist, it’s about the consumer.

It’s about as ludicrous as Robert Downey Jnr berating you for watching Iron Man on your iPad because that ‘wasn’t his vision’.

Apparently Pono is Hawaiian for righteous… wonder what the word is for ‘self righteous’? [‘Bono’, perhaps? — Ed.] The upshot? We have the technology to record, store and replay music that gives people a ‘control room experience’ if they’re willing.

It will be a war won by what makes sense to consumers, and not by the self-righteous rage of ageing rockers, the ire of studio icons, or the commercial interests of pro audio manufacturers (regardless of how worthy their intentions).

It’s possible to influence the consumer, surely? To a degree, but only if it makes practical sense. CD didn’t kill off the LP because of sound quality, it was because of convenience. And clearly, audio downloads, aren’t killing of the CD because of audio quality either. It’s convenience.

So until we can supply a better product without impacting on the consumer’s convenience, HRA will only ever lurk in the shadows.

Accounts Jaedd Asthana jaedd@alchemedia.com.au Subscriptions Miriam Mulcahy mim@alchemedia.com.au Proofreading Andrew Bencina Regular Contributors Martin Walker Paul Tingen Guy Harrison Greg Walker Greg Simmons Blair Joscelyne Mark Woods Chris Braun Robert Clark James Dampney Andrew Bencina Brent Heber Anthony Garvin Cover Image Beth Herzhaft Distribution by: Network Distribution Company.

AudioTechnology magazine (ISSN 1440-2432) is published by Alchemedia Publishing Pty Ltd (ABN 34 074 431 628).

Contact (Advertising, Subscriptions) T: +61 2 9986 1188 PO Box 6216, Frenchs Forest NSW 2086, Australia.

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E: info@alchemedia.com.au W: www.audiotechnology.com.au

All material in this magazine is copyright © 2014 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. 25/07/2014.

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COVER STORY

Black Keys’ Mastering Engineer Brian Lucey on his Process, High Sample Rates and iTunes

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TABLE OF CONTENTS

Autograph’s Andy Bruce Traces the Evolution of Musical Theatre Consoles

Studio Focus: Muso’s Corner 42

50 Unleash your Old Analogue Synths with CTRLR

Antelope Zen Studio: Get a Handle on It 52

Quick Mix: Lynard Skynard’s John Watson

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20

Shure QLX-D Wireless

Audient iD22 Desktop Interface & ASP880 Preamps

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GENERAL NEWS

TASCAM HANDHELD Tascam is weighing into the handheld recorder ring with the DR-22WL and DR-44WL. The DR44WL is a four-track recorder that includes wi-fi transport control (for remote start), with built-in stereo condenser mics shockmounted in a stereo XY pattern. A pair of XLR inputs is also available for four-track recording, and all four feed into an improved preamp and AD/DA stage. For filmmaker/ location sound people the recorder includes a hot shoe mount and a stable internal clock to avoid drifting out of sync with your camera. Wi-Fi can also be used for file transfer or streaming to smartphones or computers. The DR-22WL packs the same XY stereo mic configuration and wi-fi features but without the extra XLR inputs.

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

LEAPS & BOUNDS A couple of desirable new items from Antelope: firstly a new analogue monitoring and summing system called Satori (a Japanese Buddhist term meaning ‘enlightenment’, apparently). Satori comes in a 1U case and features 0.05dB platinum relays, Class-A preamps, phantom power, eight stereo inputs, four stereo outputs, dedicated subwoofer outlet, XLR, 1/4-inch TRS and D-Sub 25.

It also includes LED front panel buttons, talkback and level trims, analogue eight-channel summing mixer, four independent headphone outs (allowing individual feeds), and gain offset for any input/ output. The class-A preamps can also operate as Hi-Z instrument inputs and when using the MP32’s control panel, users can manipulate each of the unit’s input types and mic gain levels remotely. VUstyle metering provides instant signal confirmation by glancing at your monitor. Functionality such as mute, mono and particularly the mid-side mode allows users to work completely in-the-box.

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Next: Pure2 is a mastering-grade 24-bit/192k twochannel converter and master clock. Pure2 builds on Antelope Audio’s mastering legacy of the Eclipse 384 converter. Pure2 employs an all-new design that leverages best-in-class Burr-Brown converters and the same Acoustically Focused Clocking (AFC) technology present in Antelope’s ultra high-end Trinity master clock. When monitoring through speakers, Pure2’s relay-based stepped attenuator ensures exceptional transparency and perfect L/R balance, even at low listening levels. Additionally, Pure2’s dual-DAC technology — with separate DAC chips for left and right channels — guarantees superior stereo separation and imaging. Around the back you’ll find several wordclock outputs, S/PDIF, Toslink, AES and USB, making it suitable for nearly any studio environment.

Soundtown: (08) 9242 8055 or www.soundtown.com.au

THREE-WAY HEAVEN Neumann is expanding its KH (née Klein + Hummel) monitor range. The first of which is the company’s first midfield monitor, the KH420, an active threeway system (10/3/1-inch) used for larger 5.1 and 7.1 configurations and listening distances of up to 11 metres. Its 26Hz corner frequency is a full octave lower than the KH120. Anyone requiring even lower bass reproduction for surround systems with a dedicated LFE channel can add the KH870 woofer to extend the response down to 18Hz. The KH420 has been designed to provide dispersion regardless of the orientation of the cabinet, due to the rotatable waveguide section, which contains both the high frequency and midrange drivers. An optional Digital

Input Module (DIM1) is available, which adds digital AES3 and S/PDIF inputs and a delay function.

This feature, which is available on both the digital and analogue inputs, may be used for audio/video lip synchronisation and to compensate for nonequidistant speaker placement. Maximum delay time is 409.5ms in 0.1ms increments.

Neumann has also released a compact active threeway nearfield, the KH120, for where space is limited.

The KH120 can also be fitted with the same DIM1 module for a complete digital workflow.

Sennheiser Australia: (02) 9910 6700 or sales@sennheiser.com.au

LET THE AUDIO HUNT BEGIN These days more and more productions are selffinanced, self-produced, and out of the studio. But sometimes the quality can suffer if you don’t have good gear. Stephen Bartlett and co. are trying to do something about that with theaudiohunt.com — sort of an Airbnb for audio gear. Basically, if you’re planning a session, but you don’t have that 1176, ’Tools rig, or Neumann U87 you need to get the right result, you just jump on theaudiohunt.com and see if

anyone’s got one they can hire you. On the flipside, if you’ve got some mics or outboard gear that’s not being used daily, you can put up a few availability dates and see what falls your way. And if you can’t spare your gear, you can offer to process other user’s tracks with it. On the security side, there’s deposits and a community star rating to hep put your mind at ease, check out the FAQs for more details. The service will launch in 2015, but the site is up now.

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LIVE NEWS

NEW AGE WITH RIVAGE Yamah’s PM line of mixers has always included the company’s flagship live console. Lately, it was beginning to look like the baton was being passed to the CL line, but things are back on track with the new Rivage PM10 coming in 2015. Operationally, the PM10 includes both the Centralogic style of operation — with two large touchscreens that continue the physical channels vertically into the onscreen domain — and a Selected Channel set of hardware controls. Yamaha has also developed a new hybrid microphone preamp for its rack stage box with a more refined version of Yamaha’s ‘Natural Sound’ philosophy. There are plenty of tasty additions onboard, including a VCM model

of Rupert Neve’s Silk texture available for all the onboard preamps, plenty of Neve digital processing, TC Electronic reverbs and an impending Eventide Harmoniser to come in a future update.

Ergonomics have been overhauled, new scene and recall management features added, and close attention paid to integrating live recording — to a USB device or over Dante. With 144 input channels, 72 mix buses and 32 matrix buses, 24 DCAs and 384 plug-in slots, there’s plenty of capacity worthy of flagship status.

Yamaha Music Australia: (03) 9693 5111 or www.yamahabackstage.com.au

QSC’S AFFORDABLE DSP QSC has announced a new range of amplifiers under the GXD label. These sub-$1000 two-channel amps come in a 2U housing, and feature up to 1200W of Class D power per channel into four ohms. The front panel LCD provides a window into the onboard DSP, which handles high- and low-pass filters (24dB LR), four-band parametric equaliser, signal alignment delay, and RMS/peak speaker protection limiting. To get up and running quickly, 20 preset ‘starting points’ for selected typical systems

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are also provided. A digital limiter prevents destructive clipping while delivering the maximum usable output — protection is set by selecting the loudspeaker’s continuous power and impedance, and a choice of Mild, Medium or Aggressive protection mode as desired. On the back there are XLR and quarter-inch TRS inputs, and NL4 and binding post outputs.

Technical Audio Group: (02) 9519 0900 or www.tag.com.au

BUILD YOUR OWN POWERED BOX If you’ve dabbled in infinite baffles and find yourself pretty handy with some birch ply and a mitre box, this ‘plug and play’ amp kit from Powersoft might be of interest. The DigiMod Integration Kit is designed to give speaker builders access to a complete DSP/ amp module for an active speaker. With a couple of heatsink options to choose from, customers can also specify two different DSP solutions including interface panels with I/O connectors, LEDs, programming and networking connections. It also links in with Armonía ProManager, a dedicated

software tool for configuring the units, with the ability to render a custom-branded item of your speaker that will appear in version 2.5 of the Armonía Pro Audio Suite customer software. Installation is relatively simple, other than cutting the right size hole in your box, there are only three parts to screw together before you’ve made a fully professional looking powered speaker.

Production Audio Video Technology: (03) 9264 8000 or www.pavt.com.au

A&H PORTABLE DIGITAL CORE Allen & Heath has just dropped a mid-level expansion option into its line of live AudioRacks. Previously, the GLD and Qu line of live consoles had access to a main 24-channel digital multicore and an eight-channel expansion unit. Now there’s a 16-channel portable option too. Built more like a traditional ‘rough and ready’ stage box, the AB168 has 16 mic preamps and eight outputs, rubber bumpers on its corners and a heavy duty carry handle, though I doubt you’ll be dragging the box along the ground by its fibre cables. It connects to the consoles via Allen & Heath’s dSnake with a daisy chain connection, shows phantom power status on an LED associated with each input.

You can also get an optional rackmounting kit if 16 channels is the sweet spot for an install.

Technical Audio Group: (02) 9519 0900 or www.tag.com.au

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SOFTWARE NEWS

STRETCHING EXERCISE Pitch-shifting has been quite the trend in pop music for the last decade, ever since Cher asked if there was life after love. Of course, there’s a lot more to it than what auto-tune can offer, which is why engineers will be pleased to hear about the launch of IRCAM Lab TS (meaning ‘Transpose/Stretching’).

TS which features time stretch, pitch shifter, remix, formant, a noise tweaker with algorithms based on its unique phase vocoder engine (SuperVP) which

is said to produce accurate results, with minimal artefacts, even when used to transpose or stretch audio waveforms radically. To use the program, drag an audio file into TS and stretch it by whatever percentage you desire, then use the Formant, Transient, Sinus or Noise sliders to sculpt your sound to create new and original audio textures.

Mac only at the moment — Windows ‘soon’.

CUBASE GOES PRO With its eighth version, Steinberg’s Cubase has finally gone Pro, which may either inspire confidence or head scratches for all those ‘amateurs’ who’ve just made the professional leap. So what are these features that finally kicked Steinberg into the pro world? Well, stability took a big jump in Cubase 7, with ASIO Guard letting playback tracks render in a longer buffer so they can be pre-calculated to maximise CPU usage. Cubase Pro 8 has ASIO Guard 2, which promises even greater returns. There are a number of GUI enhancements and redesigns including: Wave Meters, which display a scrolling

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vertical waveform in the mix view to save jumping between windows; and a dockable VST instrument rack and media bay, to stop the clutter of windows.

Two other big additions are a ‘Render in place’ feature and VCA faders. The Quadrafuzz multiband distortion plug-in has had an overhaul, and new VSTs like Virtual Bass Amp, Multiband Expander and Multiband Envelope Shaper have been added. Plenty enough to warrant the jump to Pro, though there’s still the cut-down Artist version for toe-dippers.

Yamaha Music Australia: (03) 9693 5111 or www.yamahabackstage.com.au

MEGAREAPING DRUM BENEFITS Adding to the budget-conscious allure of Reaper is a new free drum sample pack developed specifically for the DAW. SM MegaReaper Drumkit is a completely free-to-use multi-sampled acoustic drum sound library, that features up to 127 velocity layers and four round robin layers per articulation, equalling an incredible four samples for each note velocity. The kit was created for Reaper by Scott McLean, then mapped by Burzukh, and is ready

to use straight out of the box. Scott really went the extra mile when recording the samples, for example the sampled snare is a 1965 Ludwig 5 x 14inch Jazz Festival snare drum using a combination of top, bottom, and stereo overhead positions.

If reaper isn’t your flavour, you can download the 2GB library and map the samples yourself, or use Todd Stillwell’s pre-formatted Kontakt template.

BOZ UNLEASHES +10DB Boz Digital Labs Black Friday plug-in release was slightly more involved than Waves’ One Knob Pumper.

Boz has released an emulation of the ADR Compex F760 compressor called the +10DB. The name also refers to the partnership Boz had with producer David Bendeth. A collector of ADR gear, he loaned a unit to Boz to model in exchange for his signature on the box.

The unit is probably most famous for featuring in old

Helios consoles, and being one part of the legendary chain that resulted in Bonham’s drum sound on When The Levee Breaks. Boz didn’t stop there though, you can also buy the +10DB bundle, which comes with the four-band EQ from the Vocal Stresser, as well as a channel strip plug-in that combines the two. $99 isn’t bad for having a virtual version of a famous compressor that’s not an 1176 or LA2A.

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BROADWING WINS Having always struggled to pin down his work to a specific genre, Ben Allen’s stage moniker, Broadwing, is fittingly universal. The Alice Springs-based songwriter, producer and occasional film composer’s songs range from electronic, to cinematic, to more guitar-based material, giving him a good background for tackling the eclectic nature of The Occupant’s track Wonderland.

Up until now, Allen’s setup has been pretty basic; comprising Reaper on a Thinkpad laptop, with a few low-priced mics, a Focusrite Scarlett interface and a relatively modest range of plugins. He mixes entirely in-the-box, occasionally routing audio through a guitar pedal.

When remixing, Allen tends not to deliberate over which parts to use for too long. Preferring to listen to the song only once or twice before jumping in. “That way I can solo a few tracks to start building my mix, and explore the muted tracks every now and then to find useful elements to play with,” he explained. “It keeps the process fresh. I don't really start out with much forethought. I just experiment and go with my instincts. The only real blueprint was to begin with a subdued feel and slowly build.”

On first listen, the element of Wonderland that grabbed Allen’s attention was “the effected vocals that weave in and out. I knew I wanted to bring them forward in my remix at certain points. I also liked the unconventional structure of the song. I wanted to keep a degree of that instead of taking one or two sections and making a three or fourminute mix.”

Thanks to:

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Allen expands on how he achieved a few of the sounds in his remix: “I wanted to try and degrade some of the backing vocals, so they were run through a particularly nasty cheap chorus pedal.

It's noisy and pretty rubbish sounding, but when hidden in the mix it was just the sound I was after.

“There’s a significant amount of acoustic guitar in my mix, although it's rarely audible. I often layer my sub bass lines with an acoustic guitar playing the exact same thing an octave or two up. It's a subtle thing but it adds some body and human error to a programmed bass line.

“There is also some pretty extreme EQ boosting and cutting. I used to be scared of making large boosts and cuts with EQ because I thought I was doing it wrong. But after reading an article where François Tétaz [one of our esteemed competition judges] talked about his use of EQ when mixing [Gotye’s] Making Mirrors, I realised there really are no rules. There are sections of my remix where the acoustic guitar has nothing going on below 600Hz. At that point it's more of a percussive, almost shaker-like element but it leaves some real estate for everything else.”

The part Allen most struggled with was mixing the elements in the buildup towards the end: “I spent far too long trying to balance everything before I realised I could just ugly it up with some meathead electric guitar playing and abandon the idea of making everything sit nicely. It's counterintuitive, but sometimes the solution to an overly busy mix is throwing more in there.”

Soon Allen will be taking delivery of an Avid ProTools HD Native system with an Omni

interface, as well as top-flight Event Opal monitors, Rode K2 valve microphone and NT5 matched pair. He’ll also be able to get some handson mixing going with an Avid Artist control surface. He’s just released his first Broadwing EP Pinhole Camera and will be using the prize to record a follow-up. He also has intentions to move further into scoring and reckons the fidelity of his recordings will improve substantially with the new gear. Allen: “I cannot overstate how huge an improvement this is to my existing gear and I am so thankful to all of the companies who supplied the prizes and AudioTechnology for running the competition.”

2ND PLACE: BEN MANUEL Sydney-based Ben Manuel is an aspiring composer/producer, but for the moment, he earns his bread teaching drums. He’s used his rhythmic prowess to good effect on The Occupants’ Wonderland, converting the verses into 7/4. It’s a world he likes to live in, challenging himself with time signatures outside the typical club-friendly 4/4. Manuel: “It made it necessary to come up with material to fill the bar completely, and the flow of the section was changed.”

Using Ableton Live rewired into Logic, Manuel was able to combine the facilities of the two DAWs. Manuel: “I created the textures at the beginning of the track by modifying and sampling the lead vocals from the chorus. The vocals were transposed onto the same note using Ableton Live’s built-in pitch shifter, and then imported into Live’s Simpler. By using its looping function and automating the sample start parameter, I was able to produce a granular effect. Scanning the sample start across the sample, and using a short loop duration, turned the texture back into discernible lyrics with their correct rhythm, while enabling me to tune them to the chord I wanted.

“In the chorus, I ran the vocals through Logic 9’s pitch correction tool and forced the pitches all onto the one note. I then

finely chopped and rearranged it all into a monophonic, but quickly changing texture, which I used as a pad. As for the melodic vocal transformation, that was simply a matter of employing and modulating Live’s pitch shift functions.

“A lot of the character of the drums in my remix comes from a heavily modified sample of the original track’s drums. I extracted a brief passage from the drum track and changed the speed and pitch relative to each other using a pitch shifter, then shaped the sound with EQ, filtering out most of the low frequencies. This gave me some percussive sounds with qualities that were not immediately identifiable as coming from a drum kit. I loaded each hit into a sampler for performance in the remix. Other drums and percussion in the track come from the TR808, some NI Kontakt libraries, and elements I created myself.”

Manuel is currently working on his entry for the Tropscore contest, and also planning a release of his own original work. Manuel reckons, “winning these prizes represents a huge improvement in the quality of my equipment, and will help me improve the quality of my work. In particular, I’m looking forward to the better monitoring I’ll get from

the Event 20/20BAS pair over my current set, as well as increasing the ease of my workflow with the Avid Artist Mix. I’ve never really had something that resembled a mixing desk before, and having something like it will be very useful.”

3RD PLACE: JOHN PARKER John Parker is a composer, performer, teacher and publisher from Brisbane, who also writes for and performs in the jazz trio Trichotomy. He’s got a lot on his plate. Add a busy family schedule, and it hasn’t left much time over the past decade for Parker to devote to electronic music. More’s the pity, because judging from his remix comp submission, he has a knack for it the years away haven’t dulled.

The remix comp was an excellent excuse to dig into an electronic Wonderland with his dusty old 002 rack and Logic Pro 9. He also uses Finale and Sibelius for his piano trio and percussion ensemble compositions.

It’s been a while since Parker last tried a remix — which happened to be AudioTechnology’s Eskimo Joe remix competition a few years ago — and he’s had a lot of time to think about his approach since then. Specifically, how to get the original out of his head. Parker: “I found it really hard to get my head out of the original last time.

I didn't want that to happen this time, so I didn’t allow myself to listen to the original track or even the stems in isolation until after I had submitted my remix. That way I'd be coming at it from a fresh perspective, while still using the source material as the basis of the remix. When I finished my remix and listened to the original it was a really spooky experience. I had spent nearly two months working on my version with 75 sampler tracks and 50-odd save files, so when I heard The Occupants' version it sounded like ghosts were coming out of my speakers!”

Parker’s remix includes a lot of cutting up and reassembling parts, as well as using samples and vocals as percussive sounds. And it all starts to get crazy towards the end of the piece. Parker: “Near the end [2:38] I used an improvised section from a little earlier in the piece [2:20], which was created by jamming out some guitar samples in five and seven groupings, in triplets. When that section was reused at the end, I pitched it up the octave and worked out the maths of how to change the source patterns into 16th notes then stretched the audio to fit — the 5s and 7s still sounded really weird.”

Parker plans to keep composing for Trichotomy and various collaborations this year.

The plan is to record another album next year, and considering they’ve always done their own pre-production, the Rode NT1 and ProTools 11 software will definitely help with that process.

“I also want to get more electro-acoustic music finished and out there,” said Parker. “So I'll be sampling more of my percussion collection at home with the Rode mic.”

AT 15

REVIEW

PORTABLE RECORDER UPDATES

RODEGrip

$40 | rodemic.com Phones have been on a steady diet since the Motorola Brick made every designer self-conscious over weight. It’s a good thing for your pocket, but our inability to get a good grip on these skinny beauties has spawned an entire industry of shopping centre phone screen fixers. This is where the Rodegrip shines. It’s an iPhone accessory for the butter fingers amongst us. The plastic handle lets you wield your phone like a pistol in use, and folds away completely to fit into your pocket. The Rodegrip comes in three flavours to match your iPhone 4, 5 or 5s, and 5c. Presumably, there’ll be iPhone 6 variants coming soon. You can also use it to mount your phone on a camera hotshoe, or sit the grip flat and use it as a mini table-top base for the mount.

A great, well-designed little accessory that’s worth carrying around if you’re serious about your i-XY. AT 16

MR MK3 STUDIO MONITORS

Mackie MR5 mk3: $199 ea Mackie MR6 mk3: $279 ea MUSIC-CENTRIC VOICING PERFECT FOR MODERN MIXING, MONITORING AND MULTIMEDIA APPLICATIONS.

ZOOM H5 Zoom jumped straight from the H4n to the H6, going from four to six inputs and solving a lot of the H4n’s niggles in the process. Only now has the numerical gap between the two been filled, and the H5 transitions the new H6 tech into H4n form factor — really, it’s a worthwhile update to the H4n.

On the shortlist of useful features aped from the H6 are removable capsules — which are interchangeable with those from the H6; dials instead of click buttons for gain — this time with bars over them, which is one step up from the H6; a power switch that won’t fall off; level setting that trims to -∞; -20dB pads on inputs 1 and 2; and backup recordings of the X/Y mic recording that are 12dB lower to cater for random peaks.

The main H4n-specific feature that has been lost is rotatable capsules — fixed at 90 degrees — you have to upgrade to the H6 X/Y capsule to get the wider 120-degree option. You also don’t have Stamina Mode, which locked you into stereo 16-bit/44.1k to conserve battery juice. There’s no need now, as you can get 15 hours out of a pair of AAs at the same setting.

While the H5 doesn’t have the full-colour screen of the H6, and is more like the H4n display. It’s rearranged to indicate the state of each channel’s low cut filter, comp/limiter, MS decoding, 48V, and pads.

The new design of the H5 X/Y capsule also has suspension rings around the capsules to reduce handling noise. And with 52dB of gain, the ability to handle 140dB SPL, and pads or backup low level recordings on inputs, the H5 will go places the H4n never could.

Price: $499 (expect to pay $379) Dynamic Music: (02) 9939 1299 or sales@dynamicmusic.com.au

RODE i-XY-L $199 | ixymic.com

Apple can sure bite back sometimes. As soon as Rode released the first i-XY, Apple did the dirty and changed its iOS device connector from the longstanding 30-pin style to the slimline Lightning adaptor. The unannounced transition hurt a lot of accessory manufacturers who lost a million potential customers overnight as iPhone 5 coveting set in. Rode always intimated it would bring out an updated i-XY, it was just a matter of how long it would take to design a sufficient support system to fit multiple phone thicknesses, and without the weight of the stereo condenser pair shearing off the slimline connector. Luckily, Rode waited long enough for the arrival of the 5c, packaging a set of shims with the new i-XY to compensate for its thinner body. It would seem easy enough for Rode to also do the same for the 6 when it’s available.

The old i-XY hasn’t gone anywhere, the i-XY-L just updates the line to cover iPhone models above the 4. It still syncs up perfectly with the Rode REC app, still has great sounding ½-inch stereo X-Y capsules with a 24-bit/96k-capable onboard AD converter to usurp the iPhone’s. The main changes are cosmetic, with a new support system that ensconces the end of the phone and slides over the home button, covering it with a button extender of its own. It also allows you to access the headphone jack on newer phones, now on the phone’s base.

My favourite bit about the new i-XY, which seems trivial, is Rode enlarged the carry case just enough to fit the i-XY in with its foam windsock. With the old one, you had to carry them separate or toss up whether to chuck the foamy in your bag. The new i-XY is everything you’d expect from an update to Rode’s successful i-XY. Still the perfect audio accessory for your iPhone if you’re serious about having a great capture device on you at any time.

Each Mackie MR mk3 Monitor Features: Clarity • Openness • Crystal Clear Mix Image • High Output • Superior Transient Response (great for snares etc) • EQ Room Controls

MACKIE DL1608 Digital Mixer with iPad Control: $1,449 PRO 16-CHANNEL DIGITAL MIXER DESIGNED TO INTEGRATE WITH YOUR IPAD FOR VERSATILE HANDSON CONTROL. 16 Onyx mic preamps 24-bit converters Built-in DSP for effects and processing.

Supports up to 10 wireless iPad connections at once (great for onstage monitor control and remote mixing).

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Shop 24/7: soundcorp.com.au AT 17

STUDIO FOCUS:

SAE MELBOURNE From the day SAE Melbourne took over its current building in South Melbourne, CEO Paul Ledingham had his eye on the factory next door. It’s been 10 years in the current digs, and in that time SAE has reached the building’s capacity by building the top floor through and adding another 1000sqm.

The one double-storey building was never going to be sufficient for the school’s growth, it needed two.

As luck would have it, on one of the rare holidays he had off, the building was put up for auction — with auction day scheduled the day after he got back. When he arrived back at work and saw the sign, Ledingham rallied the troops and in a 24-hour rush managed to do the due diligence and present a winning bid.

It looks to be an exciting time for the school, which shows no signs of slowing down despite uncertainty in government funding across the entire higher education sector. As Ledingham puts it, there’s still a lot to be said for private education, especially in niche sectors like audio and video where hands-on experience and professional interaction is a must. And while online learning supplements the course material at the moment, he doesn’t see audio education moving to solely online courses in the future. A sentiment backed up by the slew of recent hardware acquisitions.

SAE runs its audio engineering students through a series of modules designed to teach good gain structure before letting them loose on tonal-shaping devices.

The first control rooms the students AT 18

encounter are centred around simple 16-channel consoles; there’s a compressor in the room, but the main aims are to get a handle on gain structure, balancing and pan. From there, they jump up to Tascam digital mixing consoles, with a side order of hardware synths.

Then onwards to the Audient ASP8024 large format console, with dual layer control and some tasty outboard gear, and the flagship Neve Genesys room, which also ties into the best recording space in the school. The beginner suites have played host to Mackie and Behringer mixers in the past, but after having a low maintenance run with the large format Audient console, Senior Studio Technician Bradley Toan, opted to buy a couple of Audient 4816 smaller format consoles to start the students out on.

Construction is well underway next door, following the design of Phillips Smith Conwell Architects. The eventual plan will be for all the administration offices to relocate there, but in the first stage, most of the school will shift next door so the current building can be renovated.

That is likely to happen in October, with the second phase of redevelopment wrapping up in time for an April 2015 launch. Fingers crossed they’ll hit the deadline, because there’s already been a few delays. For instance, the footings for the elevator shaft were originally engineered to be four metres, so far they’ve had to dig all the way down to 26m!

When all is said and done, the flagship addition to the campus — other than a swanky new reception area and gallery that

links the two buildings — will be a 364sqm soundstage. The size of the space is easily big enough to house an orchestra. But, like most of the facilities at SAE, it will cater for all the disciplines of post production, video production, gaming, whatever course module might require a big space. To that end, it will be a well-treated acoustic sound stage, with airlock doors, a big loading dock, a lighting rig, and house one of the most versatile console to be hitting the market in its upstairs control room, an Avid S6. The S6 derives its multilayer functionality from the Euphonics S5, but supercharged. It will let the audio students feel as at home as the broadcast students, with the ability to run ProTools and Media Composer.

And if a student has put together an Ableton Live project and wants to overdub an orchestra in ProTools, they can run both sessions live on the console simultaneously. SAE has opted for the full-specced M10 master section, which means their console will also feature the waveform screens, producer section as well as banks of encoders and faders.

The main control room will be flanked by four edit suites, two on each side, each with a view of the sound stage and all tied back to the S6 when required. In preparation for the S6’s arrival, Ethernet has already been added to the tie lines in most of the existing performance areas. The plan is to invest in a rolling Focusrite Rednet rack, so students will be able to track to the Avid S6 console from anywhere in the building.

AT 19

QUICK MIX

The

with

John Watson Interview: Neil Gray

Who are you currently touring with/mixing?

FOH Engineer for Lynyrd Skynyrd.

What are some other bands that you have worked with?

My main experience and background comes from being an ex Showco and Clair Brothers employee.

How long have you been doing live sound and how did you get started?

35 years in the business. I was once a singer in a band. I wasn't too good at that so I started trying to make us sound better!

What is your favourite console and why?

Midas XL3 — great sounding preamps, EQ, etc. Some of my best shows were on this desk!

Favourite microphone or any other piece of kit?

I'm old school on mics. If the source in front of the mic is good then Shure has a mic that will work!

Most memorable gig or career highlight?

Musically my favourite artist has been Stevie Wonder. I was the monitor engineer for Stevie for several years!

Any tips/words of wisdom for someone starting out?

Don't skip Audio 101, the basics and physics of audio will always hold true.

Describe your mixing setup now, compared to what it was in 1998.

For most artists I work with at FOH and monitor positions, I always concentrate on that star vocal as the highest priority and layer things from there — but let's make that vocal larger than life!

What are three mixing techniques you regularly employ that you’ve learnt in the last 15 years?

Since day one I have been taught, and now preach and practice, gain structure with the least EQing I can get by with, but bend it if you need to! Also I’ve learnt that even though I like to build a gain structure that makes sense, sometimes you may have to veer from what makes sense depending on stage volume and your lead vocalist! I found this was particularly something to play with working for Stevie.

AT 20

Don’t skip Audio 101, the basics and physics of audio will always hold true

In the last 15 years, what are three pieces of gear or features that have come out and been game changers for you?

Of course most anyone is going to say that in the past 15 years digital consoles, digital processing, and wireless connectivity to all pieces is a major game changer in the industry. But don't forget Audio 101.

How have your working methods changed over the last 15 years?

I would say my working methods have changed as I've gotten older. I still like to be involved in the system going in and the tuning of it. But there's a lot of good systems and a lot of good system engineers out there, and when I've got a good one I let him do his job and I usually get good results. Have my mixing methods changed? From artist to artist and on different systems, yes, it changes, but I have always believed in listening to what the band is doing and to what you are doing! I don't look at the little screen all that much, it lies sometimes!

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90 Degree Studio Inside Musos Corner. 1 National Park St Newcastle West NSW 2302 02 4929 2829 www.90deg.com.au | www.musoscorner.com.au AT 21

FEATURE

Mastering engineer Brian Lucey isn’t concerned about loudness wars and high resolution files. He just wants to move air and the people that hear your mix.

Story: Mark Davie Photos:Â Beth Herzhaft

AT 22

Fortuitously, when he first took a call from ‘Dan, the producer out of Akron’ in 2008, Brian Lucey had just spent the previous couple of days mulling over the best way to answer the question mastering engineers always get asked: ‘How does your work compare with Sterling Sound, etc, etc?’

It’s a tough one. No one likes self-aggrandising, but if you’re not confident in your work, why would anyone else be either? So he gave his new answer, straight-up and honest: “Well my clients say the work is on par with those people,” started Lucey. “But obviously those are industry giants who’ve been around for decades. I will never have their history, and my name doesn’t have that kind of cachet.”

Dan got it, a real Ohio native-kind of guy; you aren’t anything special until you show it. He’d tried Sterling and the rest, and wasn’t blown away, so he started sending Lucey projects to master. The first was a record by a band called Hacienda, which racked up four stars in Rolling Stone. Lucey was excited about the positive exposure, that was a relatively big hit for a client at the time. But Dan was still just the ‘producer from Akron’ to this local Columbus, Ohio, mastering engineer.

The projects kept coming, and towards the end of an on-and-off three-week revision process for one, Lucey recalls Dan saying, ‘Well I go on tour tomorrow, so I’ll be in Phoenix, just send me the mixes whatever and we’ll talk.’ Lucey hung up the phone and thought, ‘Phoenix?!’

He explains: “In the United States of America, Phoenix is not a major market you’d think to tour, particularly a band from Akron, Ohio. From Akron, you go to Detroit, or to Chicago, New York or Philadelphia, or you go to Nashville, or Memphis, Columbus or Pittsburgh or Indianapolis.

In a 500-mile radius, you can go tour a lot of places.

Not 2000 miles to Phoenix!”

So Lucey did what anyone would do, he googled his name — ‘Dan Auerbach’ — the result was pages and pages of references to The Black Keys. Dan wasn’t just the producer from Akron, he was one half of The Black Keys.

This was pre-Brothers in ’08, though the band was already huge. But when Lucey mentioned it to Dan, he shrugged it off. After all, no one likes someone who toots their on horn.

From there the two Ohio natives kept working on material Dan was producing, like the Dr. John record, till one day Lucey got offered the new Black Keys record mixed by Tchad Blake. “I was actually having a really bad day with relationship problems,” recalled Lucey. “I’d gone to visit my parents and got this call asking if I could do the new album in three weeks. I was like, ‘yeah, I think I can probably fit that in.’”

That record ended up being Brothers, a massive hit. Lucey: “They were very successful before Brothers but I think it’s fair to say it took them up a whole new level. I was doing very well before Brothers, but I’d never done a really big record. So for both of us it was a really big record.

“It just happened,” said Lucey of the now longterm relationship. “We built trust over a couple of

years of getting to know each other. I don’t know why he initially called. I never asked him.”

Since then, Lucey has gone on to master every subsequent Black Keys release, and releases for Arctic Monkeys, Beck, Sigur Ros, Ray LaMontagne, and The Shins, as well as Aussie hits like Chet Faker’s Built on Glass.

MASTER OF MEDITATION

The path into mastering was gradual for Lucey. He worked in New York City as a professional guitarist and singer through the ’80s and ’90s, but it wasn’t until he studied Robert Fripp’s Guitar Craft that he really started meditating on the vibrational aspect of music. It was a blessing and a curse, because the more he dwelt on the mysteries of tone, the less satisfied he became with his band’s recordings.

He had a $55,000 house to his name, so he took out a loan, and started making records at home.

Eventually that led to producing, a 16-track, twoinch Ampex MM1200 and racks of outboard gear — he was getting all the tone he could eat.

Somewhere along the line he began mastering, and by 2000 it started taking over to the point of having to give up producing and being a musician.

“It didn’t take over in a way that was disappointing,” clarified Lucey. “It was as if it was

MAGIC GARDEN STUDIOS This is the third time Brian Lucey has installed a similar studio design, and he’s pretty happy with the results. He’s now based in LA, and one of the reasons he moved there from Ohio was space, which he’s got plenty of. While the proportions of Lucey’s mastering room are modest, it’s set inside the first-floor of his three-storey home. Rather than build a completely isolated space, with floating concrete floor and metre-thick walls, Lucey prefers to vent the low end rather than trap it, and let the outer open floorplan do the heavy lifting for him. It’ll even travel up to the second floor lounge, where there are guitars, a grand piano and kitchen to keep clients occupied.

Most of the treatment is focused on the mid and high frequency range. The drop ceiling is made with wooden planks instead of the usual commercial strip-metal, with four by six feet of absorption directly above the listening position to tackle the tweeters’ first point of reflection.

The absorption zone is rimmed by skyline diffusors, and outside the diffusion ring is yet more absorption that tends towards bass trapping.

The back wall is completely posable. It’s made up of six two-feet-wide, six-inch thick, eight-feethigh diffusors. The counterbalanced sections are on wheels, so he can freely move them in and out of the room if he needs the space for a listening party. But on a day-to-day level they allow him to get in behind the panels to crack the floor-to-ceiling accordion windows a touch and let a breeze filter through the space and convect up into the building. “I’ve got fresh air, sunlight and a variable back wall,” said Lucey. “And to me that is priceless.”

what I was trying to do the whole time, I just didn’t know it. To me it was about vibe. And in mastering all I do is vibe.”

Lucey learnt by making plenty of mistakes, on his own time. There was no understudy role churning out ghost-written masters for bigger names. He developed his own approach, through trial and error, a lifetime of A/B’ing and plenty of experience. And the journey has really galvanised the 47-year old’s philosophies about his craft.

IT’S ALL ABOUT VIBE

If you had to sum up Lucey’s mastering philosophies, it would be about vibe. It’s a nebulous description we use when we can’t quite put the finger on something. It’s vague and loosely interpreted, but to Lucey, the definition couldn’t be any more clear cut. For him, vibe is about vibrationally connecting listeners with the artist. It’s the only means musicians have to get the world ‘on message’, and it’s his job to ensure it sticks.

“The goal is that someone hears the single, buys the record, and goes to see the tour,” says Lucey. “At least that’s how I see it. So a single has to vibrationally connect you with the artist. The album brings in the flow factor, and I don’t think that part is any different for me than any other mastering engineer. I think a difference is the vibrational/ musical focus I have, as opposed to any specific technical thing.”

THE SIX STEPS

The trick, reckons Lucey, is to have positive compromise. “I forget who first phrased this concept, might have been Bob Olsen,” said Lucey.

“Everywhere in the production process, there’s some harm introduced to the audio. The goal is to gain more than you lose at every step.

“This might make me sound crazy, but I believe all the potential music in the world exists somewhere in this place of silence. And the artist hears and draws something from that silence and gives it a voice. At first it’s kind of raw, it might be a hook or a verse line, but it has potential.

“Then the next step is arrangement; they take that inspiration and structure it in a way that connects with more people without losing the original inspirational intent. That original rawness is lost to some extent, but they’ve arranged it in a way where you or I might get what they mean.

“You go in the studio. Let’s say it’s the new U2 record and they want to record live because they can. The result is never as cool as being in the room, something was lost, but something was gained because now we have a recording.

“To me there are six steps: Inspiration, arrangement, performance in the recording studio, tracking engineer, mixing engineer, mastering engineer. If you have a positive compromise at each step, it can be an amazing record.”

THE LOSER STRATEGY

The other side of the coin is fear, and Lucey sees it all too often in mastering, when mastering engineers dictate the final balance of a mix and mix engineers start second-guessing themselves.

AT 23

Lucey: “People tell me their last mastering engineer did all these certain things so they’re going to mix for the mastering engineer. Wrong! You need a new mastering engineer. My job is to not f**k up anything you love, while at the same time improving on everything that you love.

“It’s very LA from the old days,” says Lucey about vocal up/vocal down mixes. He personally doesn’t see the point of mixers sending him any other mix than their best one. Wonder if a vocal is too low, or bass too high? His advice is to figure it out yourself and fix it. Listen to it on lots of speakers, get the producer’s partner in for a listen and see if they like it. “You’re the production team, and the original test market,” reminds Lucey. “You should be happy with one song, done one way.

“For example, when Dan and Tchad called with the Brothers mixes, they were very respectful to me, asking things like, ‘Do you want to hear it down a dB because we’ve used a limiter?’ It was a funny thought to me, ‘Let me get this straight. You, Tchad Blake, and you, Dan Auerbach, like Mix A. And you’re asking me — who’s never heard it, had no part in the process up to this moment, didn’t write any of these songs or build any of your careers — if I want to hear Mix A down a dB? No, I want to hear Mix A. I want to hear the mix that makes you two go, ‘that’s the f**king one!’’

“Now they insisted at the time and sent Mix A minus a dB, and you wouldn’t think it mattered, but it totally mattered. It took 10 seconds to say no.

Because there’s something quite magical about that moment where a mix comes together.

“Whether you’re self-producing in your bedroom or you’ve got Danger Mouse producing your record — if you love your mix, that’s the one I want. If you start to play it safe out of fear, either making AT 24

it louder or quieter, it doesn’t matter. THE ENERGY YOU’RE BRINGING INTO IT IS FEAR. AND YOU’RE BLOWING THE SPECIALNESS OF THE SOUND YOU’VE ALL CREATED. I WANT THE SPECIAL ONE, NOT THE SAFE ONE.”

CHAIN OF TOOLS

Beyond vibe, positive compromise, and encouraging artists and mixers to stick to their guns, there is a technical aspect to what Lucey does which contributes to his success. Notably, three things he considers quite hard to do: volume without smashing it, low end without crapping it out, and harmonics that still sound clean and respectful to the source but are actually harmonically distorted.

“It’s a tricky cocktail because low end and volume tend to equal distortion,” said Lucey. “And I make no bones about it, I flirt with that line. I don’t compress much, if at all, and I can make some very loud records that still move air. It’s very important to me to move drivers. I like low end power and the intimacy that comes from it. For me, intimacy and connection with the widest audience is the goal. If we’re not moving drivers, we’re not hitting people in the chest, they’re not going to dance. It’s just going to sit there being flat, loud and annoying.”

The synergy of his signal chain is what Lucey credits as giving him the ability to flirt with that line. Lucey has AB tested the daylights out of his signal path to come to a place where the chain naturally enhances the sound, giving him the confidence to cut “hot records that move air, don’t sound like shit, and aren’t compressed to death.”

He gets in and out of the Elysia Alpha Class-A discrete compressor without destroying the musicality. And “the Pacific Microsonics A/D converter I use is going on 20 years old, but it clips

beautifully, is Class A, and has a ton of analogue headroom. There are units more perfect than mine, but I would argue that in my chain, for my ear, and for the things I front load it with, there’s nothing more musical for me.”

He uses two main EQs — a Fairman TMEQ tube equaliser, and a Focusrite Blue 315 Mk2 that’s heavily modified because “the stock version sounded like total junk. I don’t get why it was so revered. But I love the controls and functionality on it, so I had a guy named Rob Harvey go through it, gut it and see what he could do. He ended up taking out 40 capacitors and replacing all the 5534 op-amps — a super generic jellybean chip — with $16 Burr-Browns. Saying it really cleaned it up, would be an understatement.”

Lucey describes the solid state Focusrite as clean with a slightly forward quality, while the Fairman’s tubes are also clean, but it has a laid back quality: “So between the two, the image stays where it was but you get all the harmonic content.”

The last EQ he resorts to is a linear phase EQ in Sequoia, which is the first digital EQ he’s ever liked for his work. And it’s been a part of his workflow for four or five years where a minor cut or deessing is required.

He uses the hardware Waves L2 limiter, fixed to knock off a dB and a half. Again, he’s had people question his choice of limiter, but he likes the sound of it, it isn’t doing much, and recently, when the power supply went down, he just couldn’t get his chain sounding the same. “I don’t need any more than a dB and a half out of it because I’m clipping the converter,” said Lucey. “If I like the sound of it, then that’s the best thing.”

After the L2 is a Cranesong Hedd, a little trick he uses to round off the square waves with a little bit of

Two 17-inch Macbook Pros flank either side of Lucey’s listening position. They’re as big a screen as he wants in his reflection area — they sit below the throw of both the mid range driver and tweeter — and he doesn’t want any screens in front of him. He runs Seqouia on one using Bootcamp, and the other for day-to-day communication. And with such low loads on his CPU, fan noise isn’t really a problem.

He has a label on each computer, one reads, ‘Each email an honour’, and the other, ‘Each single important, every record a career’. Lucey: “People don’t have to send me an email. No matter how annoying they might be, I don’t treat them that way. It’s not my place to judge them or their music. It’s my place to connect to it.”

Whether you’re self-producing in your bedroom or you’ve got Danger Mouse producing your record — if you love your mix, that’s the mix I want

THE POWER OF TUBES Lucey uses a Cary V12 tube power amplifier to power his Joachim Gerhard-designed Canalis three-way speakers. A speaker solution he’s settled on since giving up flying the early adopter flag for Barefoot.

Although the amp has capacity for 12 tubes, Lucey runs it in a half triode/half linear mode requiring eight EL34s so he gets “some of the mid-range beauty of the triode mode but the low-high accuracy of the linear mode.

“There’s some sort of esoteric brass and maple shit going on. There are brass weights on top of the power transformer — a couple of those tightens it up, too many and it gets cold and hard. It’s to do with the magnetism of the transformer, you can hear it.

Instead of rubber feet, I’ve got it on big brass footers with little micropoints that connect to a four-inch maple plinth, and that’s on some rubber and another piece of maple. It’s in its own vibrational world.”

The tube amp front end has a couple of NOS 12BH7A tubes, “like CBS tubes from the ’50s. They’re super rare, but I just love the sound.” From there Lucey rounds out his system with current production units.

He liked the Svetlana Winged C EL34 until they stopped producing them, so he’s since switched to Chinese-made Treasure Tubes. “I don’t use NOS power tubes because that’s just stupid — both the money and consistency.”

Lucey does use NOS tubes throughout his Fairman TMEQ equaliser, for completely different reasons.

The linear phase, six-band stereo EQ has 11 tubes per side. He uses it for all his boosts, selecting pairs of NOS tubes based on the tone he favours in each frequency band: The high shelf has Telefunkens; Mullards for the high-mids; RCA blackplates for the low-mids; and either Telefunken or Sylvania down low. He’s developed his tube tastes from years as a gigging guitar player, “I use something clean on the top and the bottom and the mid-range to taste — a Telefunken will give a bright 5kHz boost, and a Mullard will be a warmer 5kHz.” One area he’s very particular about is the output section. If he doesn’t have NOS Amperex Bugle Boy EF86 tubes, he finds the whole chain goes to hell.

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harmonic distortion.

Lucey: “THERE ARE ONLY EIGHT PIECES IN MY CHAIN, INCLUDING THE COMPUTER, AND SYNERGISTICALLY THEY ALL BECOME ONE THING. At this point, it acts as an instrument for me. And my instrument is good at ‘loud’ and moving air, if that’s what needs to happen.”

MASTERED FOR EVERYONE

While the loudness wars rage on, there’s another battle going around higher sample rates, but Lucey reckons we’ve got bigger fish to fry, specifically iTunes.

Lucey: “Generally speaking, Mastered for iTunes is just about starting with the high-resolution file. But there are people using the ‘Mastered for iTunes’ label as a source of income, and I think that’s unethical, because I’ve done hours and hours of A/B testing, and tweaking the source to match the faulty codec is the wrong approach. THE RIGHT APPROACH IS TO PROVIDE THE BEST SOUNDING SOURCE AND HAVE THE CODEC GET BETTER OVER TIME.

“My hope is they improve the codec; it’s quite cold and thin. I don’t think the guy writing the code hears the way some of us hear. Some people don’t hear harmonic content as being particularly valuable, and the guy at Apple is going for a perfectionistic approach. On paper, the codec is much closer to the source than an mp3 at 256kbps — mp3 is more distorted, no doubt. But in my opinion there are many things about an mp3 that sound much better.

“Distortion on paper is not the enemy, everything we do is distortion, it’s part of the alchemy — frequency, distortion, transients. WHEN I LISTEN TO MUSIC THROUGH THE APPLE CODEC I HEAR THINNER LOW END, SHORTENED BASS NOTES AND IT’S HARMONICALLY COLDER. WHEN I LISTEN TO AN MP3 I HEAR THE OPPOSITE, FUZZY AND BLOATED. It’s like a hippie

that’s gone four days without a shower and shows up at the party. But there are some qualities to that we could really use in Apple’s codec, which has a real white labcoat, ’80s CD aesthetic. Let’s hope Apple improves, or we go to streaming 24-bit.

“As for the whole Hi-Def movement, that’s a whole other silliness, upgrading and printing something at 96k is silly. Converters sound like they sound. A converter has three components: the actual chip; the analogue circuitry that leads from the input/output to the chip and back; and the clock that runs the process. THE USER INPUT TO THOSE

THREE FIXED THINGS IS SAMPLE RATE. BUT THOSE THREE THINGS NEVER CHANGE FROM 44.1K TO 192K. YOU STILL HAVE THE SAME CHIP, CLOCK AND ANALOGUE CIRCUITRY IN THE PATH. By changing the sample rate, you’re simply

changing the presentation of those things.

“It’s like mastering, I could master a record five ways, but I wouldn’t change the song, the artist or the mix — I might make it a little brighter, midrangier, darker, fatter, or a little louder or softer.

I would still just be playing with this fixed thing, which is the mix. So when you’re chopping sound up into more slices, all you’re doing is playing with the frequency and depth presentation. Now the depth part of it, 99 times out of 100, it doesn’t matter because you don’t have enough dynamics in the first place for it to really matter.

“If you’re talking about recording a symphony,

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AT 26 dPg_AudioTech_Jun14-out.indd 1

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okay, but I’d still rather have an awesome converter running at 44.1k. Any asshole can print at 192k with a $150 converter, is that better than my $17,000 ‘development cost was no object’ converter running Class A at 44.1k?

“This seems pretty simple to me. There’s a whole economy springing up over high sample rate delivery — it’s bullshit. Look at all the records Daniel Lanois recorded to a DAT machine.

Wrecking Ball with Emmy-Lou Harris sounds great, but was mixed to DAT. By the Apple iTunes standard that can never be mastered for iTunes because the source is 16-bit. That’s dumb! Some jackass can send you a 24-bit source of the worstsounding mix in the world, it doesn’t make it a better record. The numbers game is very male, like a version of penis-measuring, it’s not important.

“Yeah, you hear more high frequency detail, more reverb tail. But it’s still those three components that matter most. Stream at 24-bit and be done with it. The 16 to 24-bit change is nice, after that it’s kind of silly.”

In the same way that Lucey doesn’t think about codecs when mastering, he pays no heed to alternate listening systems either. “Tighten Up, the single on The Black Keys’ Brothers album has been licensed more than any other song in The US.,” said Lucey. “So I’ve heard Tighten Up everywhere — bars, grocery stores, strip clubs, car after car, and show after show on TV. And no matter where I hear it, it still sounds how I meant it to sound.

It still has good low-end and good low-mids, it’s still loud, it’s clear, it’s still harmonically nice, even when it’s been data compressed down to whatever shitty format they’re using. I don’t make a thing less bright, so if it gets brighter it’s okay, and I don’t make a thing fatter, so if it gets thinner it’s okay. I don’t worry about that.”

CONNECTING WORLDWIDE

Mastering engineers are the midwives of the audio world, controlling the moment between gestation and becoming real in the world. Lucey doesn’t take that responsibility lightly, he knows once it leaves his hands it will never be touched again, and the world can judge it. It’s a big part of the reason Lucey has foregone the “whole notion of a mastering room being a $150,000-concrete foundation, multiwalled layered cave” and persevered with his own non-isolated mastering room design.

“To me this is what mastering is,” he reasons.

“I’m at the threshold between the creation and the world. I can literally look out and see the world go by. I’ve done the windowless cave, perfect room, and it’s not my aesthetic. My aesthetic is naturalism and connection.

“When you get into a scientific mindset of perfectionism, you end up with an amplifier that has a million details into the horizon line, but not the palpability of the mid-range, which is where the people are. Details are ridiculous, no-one listens to details. They listen to music and music is mostly mid-range.

“So my setup — from the location, to the amplifier, to the way I think about it — is all about that connection, because there’s not enough humanity in the windowless cave.”

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FEATURE

FROM THE TRIDENT FLEXIMIX TO DIGICO SD7T:

MIXING MUSICAL THEATRE Andrew Bruce, owner and founder of seminal West End specialists, Autograph Sound, charts the history of the mixing console in musical theatre.

Story: Andrew Bruce

We didn’t realise it at the time, but mixing on a digital work surface was something we had been slowly working towards since the very start of our company — a subconscious vocation, if you like. It took a while, but we gradually built up a list of indispensable features for the ideal theatre console.

1973

1976

1981

1973: WHERE IT ALL BEGAN

1976: TRIDENT FLEXIMIX

1981: MIDAS TR

When we first started in 1973, we had to adapt products that patently weren’t suited to theatre in order to make them useful. To start with, we were mainly doing plays, but coming from the Royal Opera House [where Andrew was head of sound at the time], music played a very big part in our lives and gradually we started doing more musical projects. In the course of the first 15 years, we accumulated a group of features that were absolutely imperative for theatre.

In 1973, there were no loudspeakers, mixers, radio mics or miniature mics that were made specifically for theatre use. And there were only three legal radio frequencies we could use. We had to learn how to persuade manufacturers to see things our way, which was almost impossible to do because we had no money to commission anything — manufacturers’ first question is always, “how many will you order if we put time into this?”.

When we were awarded our first musical contract — we supplied the equipment for A Chorus Line in 1976 — Abe Jacob [Broadway sound designer, ‘The Godfather of Theatre Sound’] had chosen a console we otherwise would normally have had no prospect of being able to buy ourselves, but in this case we were protected, because we had a guaranteed income from the production.

That was the very first Trident Fleximix off the production line, which was a very early and quite rudimentary modular mixer aimed at the live market. It had one particularly good feature — you were able to choose what overall size of frame you wanted from a selection of different sized stock items and bolt them all together. It was cheap(-ish) and cheerful and you could mix and match modules in the frame so you weren’t limited to a fixed layout of channels and groups.

The main attraction was the modularity and flexible frame size.

We did a series of shows for Abe using the Fleximix including the original Evita, but it was Cats in 1981 that proved to be the next significant milestone. It was a big step up, a real mainstream Andrew LloydWebber musical with parts of it dizzyingly close to rock ’n’ roll; Abe had chosen a Midas TR console, which was originally designed for the National Theatre and had an output matrix in it. We had modularity, now we had a theatre matrix…

Nine years in and by this point we were getting big enough to commission our own consoles.

Not long after Cats we did a show called Song & Dance at the Palace Theatre, which was another Andrew Lloyd Webber show. Julian Beech and I designed it, but Andrew insisted that Martin Levan — his engineer from Morgan Studios — mix it.

We had put a Trident Trimix studio mixer on the show, and it wasn’t great — we were following a path we knew wasn’t right. Martin knew Clive Green from Morgan Studios, because they had a big Cadac desk. I also knew Cadac from my time at the Opera House as they were one of the choices when we were in the market for a proper console — but they didn’t make anything small enough, and we chose a Neve instead. What I didn’t know was that they’d gone out of business in the meantime. Martin suggested we pay Clive a visit because he knew Clive had bought back the remnants of the company and was hungry to find a new niche market.

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1983 1983: CADAC ‘THE COFFIN’

STARLIGHT EXPRESS ONE-OFF

PROTO SCENE AUTOMATION

This coincided with a show that Martin had been asked to design — Little Shop of Horrors, which was going into the Comedy [now the Harold Pinter] Theatre — and we had a very specific brief that we couldn’t take out more than four seats. We needed to commission a console that would fit into that space, and Clive agreed to build it. It was conceived, designed, built and delivered in four weeks from scratch.

The only things he didn’t deliver in the time were the power supplies, so we ran it on bench supplies for a while. We still have that console today. It was christened The Coffin, but it was a brilliant little mixer that sounded absolutely extraordinary; we now realised there was a manufacturer who saw the theatre as a recognisable market.

By now, we had also established a relationship with John and Helen Meyer through Abe, who had come out with the UPA loudspeaker just in time for Cats, which was turning out to be an ideal theatre speaker. They knew the market was there for developing theatre speakers — small and powerful because Victorian theatres often have small proscenium arches where traditional loudspeakers just don’t work.

So, by this time, there were two manufacturers who were taking notice of theatre in a big way.

The next milestone for us was Starlight Express; it was Martin’s second show and was a huge leap in size and complexity. WHAT WE DID MAKES MY

It was 1983 and shows were becoming complex enough that the actual operation required some kind of assistance, and computers were reasonably commonplace.

Clive Green commissioned a chap called Derek Dearden, who was a professor of computing at London University, to build a rudimentary computer that could remember control group assignments. You could program a series of cues that brought the elements you wanted to the central VCA section, so the operator wasn’t scooting up and down a huge console trying to find things in the dark. That was the beginning of a real sea change in the way mixing in the theatre developed; the idea that things should come to you — and that required computer help.

Meanwhile, we had started a program with Clive to design a series of standard consoles for the hire company [Autograph] that could be used for any musical. This idea of building consoles specifically for each show couldn’t last forever. LUCKILY, STARLIGHT LASTED 21 YEARS SO WE

BLOOD RUN COLD TO THIS DAY — WE COMMISSIONED A CONSOLE THAT WAS A COMPLETE ONE-OFF AND HAD NO USE WHATSOEVER OUTSIDE OF THAT ONE SPECIFIC SHOW. It consisted of two consoles in one; it was a vocal console that only routed to two buses, and an orchestral console that routed to many more with the equivalent of a monitor matrix on every channel — using screwdriver pots rather than switches to route to the groups.

It was designed to fit with the concept of the show, where the back wall of the stage was to be composed of just loudspeakers, with every loudspeaker being a different instrument or section of the band. It didn’t quite work out that way, but it came close. The console cost a fortune and, if the show had folded, we would have been stuck with this monumental white elephant. It did have another thing, however…

GOT OUR MONEY BACK. INCIDENTALLY, THE COMPUTER — JUST A CARD WITH WIRES HANGING OUT OF THE BACK OF IT — ONLY FAILED ONCE DURING THAT TIME.

1985 1985: CADAC A TYPE

So we had started to develop this range of consoles with Clive that embodied all the various things we had learned up to this stage — modularity of frames, output matrix, balanced bussing, all those things that were, by now, deemed totally indispensable. The simplest of these was the A-Type console, which was designed without any automation as that was still a bit experimental and added an unwelcome level of expense. We put the first

one on Les Misérables in 1985.

That particular console ran eight shows a week for 15 years before being replaced by an automated J-Type two years before the show moved from the Palace to the Queen’s Theatre in 2002. While the A-type was totally manual, the J-Type that replaced it for the last two years had VCA assistance from the computer, which instantly simplified a very busy three-and-ahalf hour mix.

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1989

1992

1989: CADAC E TYPE

1992: SOUNDCRAFT BROADWAY PROGRAM

In the intervening years, we had commissioned the next series, the E-Type, from Clive, which had expanded facilities — VCAs and programmable control groups became a standard feature so we could start to automate properly. We started developing our own software to control the assignment of control groups, but Clive also had his own versions most of which, I’m bound to say, were much inferior. Developing our own software allowed us to expand on various extras, like having active legend strips sitting above the control group faders that displayed the names of individuals or groups of people who were assigned to them on a cueby-cue basis.

It was an everexpanding brief and widening set of features that we were working on over the years. Miss Saigon in 1989 was the first show that had the legend strips on an E-Type, so it was all coming together.

Nevertheless, these were still big old analogue consoles that were taking up more and more space.

Cameron Macintosh in particular wasted no opportunity in pointing out how much money he was losing because of the number of seats we took up, and could we not work towards the same sort of thing that the automation and lighting guys/ gals had done? It was an interesting point — but we were struggling with the old analogue/digital debate: which sounds best?… not something that relevant to automation or lighting, and which only the most astute producer would understand.

But we were definitely at the point where we were having to argue the toss with Cameron and other producers saying, “analogue is still absolutely the only way to go and analogue requires space.

There’s no assignability; we can’t bury channels and bring them to the surface, they all have to be there simultaneously. It does mean the operator has to do a lot of moving around during the show fixing things, but right now there is no viable alternative.”

We started having to dig our heels in with the producers, who were the ones giving us the jobs, so we had to look for counter arguments, my favourite of which was: “Well yes, it’s all very well saying you’re losing money because of the seats we’ve taken out but that only applies if you’re 100 per cent full all the time, which is not necessarily the case.” Sadly for my argument, it was exactly the case with most of Cameron’s shows at the time.

As a result of this relentless pressure from producers, we were constantly on the lookout when the first digital consoles started appearing.

We listened to them and tried to make sense of their user interface from our rather specialised point-of-view then, without exception, rejected them for our own use because of their lack of features and their sound. We continued down the route with analogue and we had a lucky break when I was approached in 1992 out of the blue by Chas Brooke and Nigel Olliff, who were both then with BSS, to see if I was interested in joining in a development program with Soundcraft.

In 1973, there were no loudspeakers, mixers, radio mics or miniature mics that were made specifically for theatre use. And there were only three legal radio frequencies we could use.

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Soundcraft had decided that theatre sound was really worth taking notice of, and approached Chas and Nigel to find people in the industry who could help them define the feature set — both hardware and software — that should be included in this new assignable console. It was very, very enlightened of them at that time. I jumped at the idea because we knew we had to talk to somebody and it would be great to be there on the ground floor to influence the hardware, as well as the software features. So we had a series of meetings with the team they had put together to simultaneously define and develop the software and hardware.

It gave me an opportunity to really sit down, with the help of everybody here [at Autograph] and several people outside the company, and to think through what we needed in terms of software and particularly the layout of controllers we would want to see on the surface while keeping it small, but not too small. More than two years of work went into that — and it was quite detailed. We got into some genuine blue-sky thinking about the sorts of routines we wanted to incorporate into this clever new digital console that had never been possible before and would make our lives a whole lot easier.

It was really the chance of a lifetime to make a difference for theatre sound.

ABOUT ANDREW BRUCE Pioneering theatre sound designer Andrew Bruce is the founder and owner of Autograph.

He began his career at the Glyndebourne Festival Opera before becoming Head of Sound at the Royal Opera House, Covent Garden. His extensive musical credits include Evita, Cats, Les Misérables and Miss Saigon.

For about six months [we] did do some work on a first-version digital Cadac console … but it wasn’t long before he said, “we just don’t have the money required to do this”

CELINE: OFF BROADWAY

1996 1996: BROADWAY PROTOTYPE

I then had an opportunity with Cameron — who had never let up with the pressure to downsize — to wheel out the very first prototype on a musical called Martin Guerre, which he produced in 1996. This was the perfect test-bed because I knew Cameron well and was able to say to him, “we’ve finally got something we’d like to try… we’re not entirely confident about it because it’s very early days but we’ve been working on it with a manufacturer for nearly three years. How about we try putting it in on your new show? But, just in case, we’ll put an old Cadac in as well for technicals until it’s proved itself, so if it fails we’ll have something to fall back on.” He said, “fine, let’s try it.”

So we did, we installed the first Soundcraft Broadway at the Prince Edward theatre in the stalls on the right-hand side of the aisle with a Cadac J-Type on the left-hand side of the aisle at the Prince Edward. Which unfortunately meant that we had to have two completely separate cable infrastructures because all the racks for the Broadway were under the stage, while there was a complete alternative set of wiring to the Cadac in 1 10/06/14 the front of EMA_AT102_[Press}.pdf house.

Ultimately, the Broadway never actually controlled any of the audio on the show. It fell over so frequently that, almost on the first day, we changed to the Cadac and never changed back. We let them stay there while they systematically tried to bug fix it, but the networking technology of the time was far too slow. It could successfully handle a single microphone, even as many as five, but once even moderate amounts of data started flying around the network between the stage racks and the front-of-house surface, it fell over instantly.

It was very depressing, particularly for them but also for us, because we were interrupted in our development process and we had to turn all of our attention back to doing things the old-fashioned way. Nevertheless, it had got the ball rolling and after they took it out, they continued development assuring us they had learnt a lot of valuable stuff.

That was until the new and incoming head of Harman came to see me one day and said, “I’m afraid we’ve decided to knock it on the head because it’s taking up too much development time and resources, and we’re actually falling behind the other manufacturers because they have all started to develop smaller consoles with limited digital implementation and automation.”

11:40 AM

The Broadway had gone on to do a Celine Dion world tour before gracefully fading away. I lost touch with them after our aborted attempt so I don’t know how they managed to get it to that point, but good on them for persevering.

I think they learnt the networking was the bottleneck. I can’t remember what the networking speeds were, but it just couldn’t keep up. The hardware was beautifully made and, although we very rarely heard anything out of it, the digital control of analogue seemed to be fine and the surface seemed to work, it was just that the exchange of information couldn’t keep up — by a factor of a hundred.

The work that had been done remained valid though. We had identified basic principles, sorted them out in our minds, re-evaluated them and then passed them on to the software writers to interpret.

WE WERE FORCED TO THINK ABOUT EXACTLY WHAT WE DO AND HOW WE DO IT, AND THE ORDER OF THINGS AND THEIR PRIORITIES. Indeed, at the end of the Soundcraft

project I was left with two folders of information, which I knew would be useful one day.

After that, we returned to Cadac and Clive with our tail between our legs. He knew we were under increasing pressure from producers to come up with something smaller (and therefore assignable), and he knew he couldn’t deliver. Nevertheless, he said, “come on, let’s start on a digital console.”

For about six months, he and I — using the same information, because I hadn’t been asked to sign any non-disclosure agreements — did do some work on a first-version digital Cadac console. And some good came out of that, mainly hardware layouts, but it wasn’t long before he said, “we just don’t have the money required to do this”. So, reluctantly, we shelved it again.

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1999

2002

1999: CADAC SAM SOFTWARE

2002: GETTING TO KNOW DIGICO

We continued designing shows using the J-Type, which was his latest analogue console. Simultaneously, we had continued with the development of our in-house software to control the E and J consoles but then Clive, for some inexplicable reason, decided he would no longer share the code with us for the latest consoles and we hit a brick wall. So we were forced to abandon the development of our in-house console software and adopt his, which was called SAM. We did some work with Philip at Cadac on this and many elements — visuals and functionality — became clearer as did, in some cases, their undesirability.

This only further refined our set of essential features that had to be in a digital console. SAM was first used on Mamma Mia, also at the Prince Edward in 1999.

2002 was the year Digico first showed its D5 Live at the PLASA show in London. I remember being told by a colleague, Bobby Aitken, that I should go along and take a look, because this was the next step in digital consoles. At this point we’d pretty much written off anything that existed in the digital console world as being not much use for musical theatre. We understood the direction that conventional software had taken but, to be quite honest, I didn’t understand how the system of libraries and scope could possibly be relevant to us because the way we work is very, very specific and is quite unknown outside of theatre.

I hung around at the end of one of his demonstrations, took a look and said [to Digico MD James Gordon] that I’d done a bit of work with Soundcraft and Cadac, and had a mine of information on an application they may not have considered.

Would they be interested in evolving something specifically for the theatre?

FINALLY: DIGITAL DREAM COMES TRUE Recognising the D5’s theatre potential, Andrew Bruce called in as many sound designers and operators as he could to gather their input and see if they shared his enthusiasm. They did, and their thoughts added to the inspiration behind what was to become the ‘T’ version software.

“There was very little time for beta testing before we had the opportunity to make the first foray,”’ Bruce says. “This coincided with the move of Les Mis from the Palace Theatre, where it had been for 19 years on an A-Type Cadac, to the Queen’s, which is a much smaller theatre and the producer needed all the seats he could get. It was a perfect opportunity because it was a show we knew very well — we knew where all the cue points had to be and we knew how it had to sound.

“On the day of our first preview, we had two software updates — one in the morning and one just after lunch, to get rid of a major problem we were encountering. It was a scary day, but we did the first preview, with no problems.

“The fact we could phone through and describe what

AT 32

was happening — they got it, they altered it, they sent it back to us and we loaded it, all in the course of one day — encapsulates Digico to me.”

What began as additional software for the original DiGiCo D5 console was to take a significant step forward with the subsequent SD7 desk.

Bruce: “All the theatre stuff in the D5 was bolted on to the core software, routines that made the D5 work in a different way. Some were totally successful throughout their life, but some made interrogation of programming very challenging. We quickly learned to treat the most troublesome ones with great care.

The most risky routine was for external programming of control group membership.”

With the SD platform, all of the things that Bruce, the other sound designers and Digico had developed were an integral part of the desk’s software, including access to a session at a much deeper level than on the D5. Among other things, this allows on-line access to a show’s session programming without interfering with a performance, using an additional surface or computer. — Tim Goodyear

AT 33

TUTORIAL

CTRLR:

The Missing Link?

CTRLR integrates hardware synths into your DAW with the convenience of software synths.

See you on Ebay.

Column: Jason Hearn

Around the year 2000, I was lured by the spoils of software synths: reliable recall, sounds stored within project files, easy automation and the ability to run multiple instances. It was revolutionary. But ever since, some downright gorgeous-sounding hardware synths have been slowly collecting dust around me, exactly because of this lack of convenience.

Just one example is a Roland MKS80 Super Jupiter polyphonic analogue synth module from the mid ’80s I’d been loaned. Since I didn’t have the optional MPG80 programming unit for it — with all the knobs and faders — it sat under-utilised in my rack. When another good friend finally loaned me his MPG80 controller, what should have been the connecting link still didn’t solve two aspects of soft-synth workflow I’d become accustomed to: full recall of sound parameters from within my DAW projects (I’m using a DAW which doesn’t record SysEx dumps), and full automation of all parameters.

It got me thinking. What would I need to build a VST/AU-based editor for my synth collection?

When I started listing the skills required, even an AT 34

optimistic timeframe was starting to look medical degree-long: Learn how to code C++; build GUIs from scratch; create VST/AU plug-ins; ensure it was cross platform; and test it. It sounded less and less like a fun weeknight hobby and more like a career choice.

Thankfully, almost out of the blue, I discovered CTRLR, which has completely revolutionised the way I integrate my hardware synths within the context of a modern-day 64-bit DAW. Hopefully it can do the same for you.

WHAT IS CTRLR?

CTRLR (ctrlr.org) is a freeware, open-source, cross-platform tool allowing owners of MIDIequipped hardware synths to integrate them with the same convenience of software-based plug-in synths. It can run either as a standalone editor or a VST/AU plug-in within 32-bit and 64-bit DAW applications running on OS X, Windows, and surprisingly, Linux.

Launching CTRLR as a VST/AU plug-in allows you to load a controller Panel created for your synth, configure the MIDI port and channel

settings, and then, via standard plug-in automation, automate all parameter objects on the Panel just like a software synth.

While the concept itself may not be unique (manufacturers of modern synths often provide a plug-in-based editor), what is special about CTRLR is users are free to create their own custom Panels with relative ease — providing it has a sufficient MIDI spec.

Additionally, CTRLR allows you to generate specific VST/AU plug-ins with your desired Panel preloaded. I’ve spent the past months getting a kick out of addressing the Roland MKS80 nestled deep within my rack by simply calling up my self-created plug-in called Roland MKS80.vst.

A FEATURE-RICH EDITOR

CTRLR’s editor is deep to say the least. Here are some examples: • You can arrange layers of controls in Tabs for control-laden synths.

• Users with design chops can import bitmaps as the basis for their GUI and replace CTRLR’s built-in faders and knobs with bitmap animations

saved in PNG format. The freeware KnobMan application (www.g200kg.com/en/software/ knobman) is designed specifically with this type of design in mind.

• CTRLR’s slider/knob objects even support multi-part MIDI messages — great news for synths supporting high-resolution 14-bit Continuous Controller data such as Moog’s Voyager. And when creating Panel Modulators that send SysEx messages, complexities such as Roland’s SysEx Checksum standards have been implemented (this was essential when I created the Roland MKS80 panel).

• There’s also a very useful MIDI Monitor tool allowing you to watch the data flowing in and out of CTRLR. And the MIDI Calculator window will help those not accustomed to converting values back and forth from decimal to hexadecimal, and vice versa.

TOTAL RECALL

Prior to making the MKS80 Panel, external modulation via MIDI was limited. Not anymore. Okay, we can all dust off our Roland Super Jupiters now.

CTRLR PANELS CURRENTLY AVAILABLE FOR DOWNLOAD: Access Virus TI, Akai MiniAK, Alesis Micron (same panel as Akai MiniAK), Cheetah MS6 with MAD Rom panel, DSI Mopho, DSI Prophet’08 (Power’08), Emagic AMT8, EMU Procussion, GEM Real Expander, Jomox Xbase09, Korg DW6000, Korg DW8000 (two different panels), Korg Prophecy, MFB SynthLite 2, MIDIbox SID, Moog Little Fatty (Mogue Big Skinny), Moog Minitaur, Moog Voyager (Mogue Voyeur), Novation Supernova, Roland Juno

106, Roland JV/XP family panel (Jay-V) ,Roland JX-3P fitted Kiwitechnics 3P Upgrade (Kiwi-3P), Roland JX3P with Organix MIDI Kit (JOX’3P), Roland JX8P (two different panels), Roland MKS80, Roland VM-3100 Mixer Map, Sequential Circuits Six-Trak (Six-Trak Editor), Studio Electronics Omega 8, Vintage Revolution PedalPro MIDI, Waldorf Microwave II/XT/XTk, Waldorf Rack Attack (RealAttack), Yamaha TX7, Yamaha TX81Z.

While building your own panels is powerful, the key feature most of us are looking for is the ability to recall our hardware synth settings as easily as our softsynths. So, how does it work?

Well, behind the scenes CTRLR’s primary developer, Atom, is hard at work building a Librarian within CTRLR that promises to help manage the patch memories within your hardware synths. It’s a big job, so in the meantime CTRLR uses the Snapshot function to achieve total recall of patches within DAW projects — perfectly adequate if you tend to create unique sounds for each project you create.

Creating a Snapshot stores a ‘local’ preset in that project’s plug-in instance. When you reopen the project, simply hit Send Snapshot, and CTRLR transmits all settings on the Panel to the synth. Hey presto, your hardware synth sound is recalled.

Used in tandem with your DAW’s Freeze function, you can use multiple instances of a single hardware synth just like you can with a software synth. Just unfreeze and resend your snapshot if you want to tweak the sound. The best part of this workflow is the sounds are stored within the DAW project itself (just like a software synth), freeing you from having to separately maintain patch memories within the synth itself. It also effectively gives you unlimited patch memories for your synth — one of the key motivators for me avoiding hardware synths in the first place.

Editing is uni-directional (you can’t actually suck the patch data out of the synth to edit) unless otherwise stated. This is fine for most purposes if you create yourself a good initial Snapshot.

PACKETS OF STREAMING GOLD

Some synths only adjust parameters with MIDI SysEx messages, which is a problem for Ableton Live users as the DAW removes MIDI SysEx messages from MIDI data streams. CTRLR neatly overcomes this limitation by communicating directly with hardware MIDI ports (in Live’s MIDI preferences, uncheck the Input and Output ports you are using with CTRLR for the Track/Sync/ Remote columns). Although it has been possible AT 35

for Ableton Live users to create Max4Live editors for hardware synths, for synths requiring SysEx messages you need to launch a satellite Max run-time application when recalling projects, introducing the risk of incomplete recall.

When creating a CTRLR Panel for a synth without the aid of the manufacturer’s MIDI SysEx Implementation guide, consider watching the data streams from an existing editor like Emagic’s Windows-only SoundDive, and identifying which values are changing in response to tweaking the controls.

A similar approach was instrumental when I developed the MKS80 Panel. Armed with an iPad2 running Liine’s Lemur app loaded with the MKS80 editor, I watched the data streams — making it easier to understand the grubby scanned version of the MKS80’s MIDI SysEx guide I found online.

SUPERCHARGE SYNTHS WITH CTRLR

Prior to making the MKS80 Panel, external modulation via MIDI was limited to Aftertouch (with only two destinations), Velocity (rates for Envelopes 1 and 2 only) and Pitch Wheel. Now, armed with the Panel, any parameters within the MKS80 can be modulated. In tandem with Ableton Live’s Clip-based parameter automation, I can perform exotic modulations such as ‘wavesequencing’ oscillator waveforms. It’s not just controlling, it’s super-charging your synths!

Creating your own Panels might seem daunting, but it’s doable and there’s a whole community to help you along. And during the process you’ll get a dual education — how to build Panels, and a deep understanding of what makes your synth tick — all without the medical degree timeframe.

AT 36

POWER TIP: SET YOUR PRESETS If your DAW has track-insertable objects to bring external audio streams into a track (for example, Ableton Live’s External Effect object) and the ability to save track presets, you’re in luck! With these facilities, you can simultaneously launch an instance of CTRLR with MIDI settings preconfigured and a Panel preloaded, simplifying how you route audio streams back into the DAW. Much more convenient than manually creating audio tracks to route your synth’s audio each time.

This is a huge timesaver and great for keeping in the creative zone. Ideally, you will be using an audio interface with enough I/O to dedicate specific inputs to specific synths and keep them connected consistently. In fact, the existence of a good CTRLR Panel for a forgotten classic on eBay is good enough reason to consider buying it.

POWER TIP: GEEKING OUT ON LUA For ultra-geeky musicians familiar with programming and LUA scripting, CTRLR’s Panel editor has a full LUA scripting engine and console built-in. It allows you to build and attach LUA ‘Methods’ to modulators that execute depending on specific GUI actions. For most Panel creators, understanding LUA scripting is unnecessary. If, however, you wish to extend a Panel to interpret SysEx patch dumps from your synth or build additional functionality into your Panels such as Patch randomisers, LUA will be worthwhile learning.

POWER TIP: THINK BEFORE YOU CLICK In Ableton, parameters of a softsynth are not made available for automation until you click the Configure tool (found in the header of the plug-in device container) and then click on each parameter on the synth’s GUI until all are shown. It’s worth putting some thought into the order you add the parameters since this is the order they’ll be presented and grouped on your control surface.

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AT 37

TUTORIAL

CAN YOU TELL THE DECIBEL? Coach Covill gets back to basics with a decibel primer, drilling down to the types you should keep your ear out for.

Tutorial: Hugh Covill

The decibel is the common language of audio folk but many remain unsure how to speak this language fluently. There are a plethora of decibel dialects you’ll come across on your audio journey. Here’s just a few off the top of my head: dBm, dBW, dBV, dBmV, dBuV, dBu, dBuA, dBHz, dBSPL, dBA, dBTP, dBSWL, dBSIL… The list goes on and on. In this primer I’ll break decibels down to their nuts and bolts, provide some ‘rules of thumb’ for their usage, argue which flavours are the most essential, and for those who are maths averse, all without citing a single logarithmic equation. Hopefully we’ll traverse the yellow brick road of volts, watts and sound pressure levels to emerge adept and fluent decibel linguists.

WHY THE DECIBEL?

The decibel came out of Bell Laboratories in the 1920s, hence the Deci‘bel’. The ’20s saw an exponential adoption of telephone use and engineers needed a methodology and scale to quantify signal loss in telephone lines.

THE AT DECIBEL AIDE

AT THAT TIME, THE DECIBEL MEASURED THE CHANGE IN POWER OVER A MILE OF STANDARD CABLE WITH DEFINED PROPERTIES. By 1940, the VU meter (using a

decibel scale) was adopted across the telephony industry and beyond to broadcasting and recording — the decibel had arrived.

WHAT IS A DECIBEL?

A dB is a logarithmic unit. It describes the ratio (change) between two values. In audio these values are usually power or voltage. Now I promised there would be no maths so I won’t elaborate further on the empirical definition. The key things to understand are: • A dB is a dimensionless unit in the same way a percentage is.

Percentages are used to express how large or small one quantity is relative to another quantity. Decibels describe the change from one volume/ voltage/power state to another.

• The dB uses a logarithmic scale. Logarithms are just a way of

micropascals

dBSPL 170

Hopefully, this chart will bring these decibel concepts into focus.

Travelling from left to right will take you through how the decibel is used from input to output, with the very far right scale demonstrating subjective listening responses to changes in level.

150

Analog Audio Power Amps

Jet Engine 1m

140

130 Digital Audio DAWs, Sound Cards, Audio Interfaces

Stun Grenade

160

200,000,000

Analog Audio Mixers, Microphones, Compressors & Processors

Comparative Sound

Threshold of Pain

Acoustic Domain 120

110

Thunder Clap

100

Night Club

90

Heavy Traffic

dB Watts 0 CLIP

24.5

dBu 30

20

2.45

10

dBW

2,000,000

Clip Peak signal level

200,000

80

63,245

70

Twice as loud

10

9

8

Line Level +4dBu 0

-10

77mV

-20

24mV

-30

Consumer level

Vacuum Cleaner 1m

60

6324

50

100,000

50

10,000

40

40

1,000

30

30

100

20

10

10

7

1.5 times as loud

Quiet Street

6

5

-20

240mV

Nominal signal level

-15

0.775

-10

-6

7.75

Console max. output before clip

-3

Volts

2.4mV

-50

Microphone level range

-50

-40

-60

7.7mV

-40

-30

4

dBu 0dBu=.775v(ms) ref.

Absolute.

AT 38

Noise floor

1

0

200

Quiet Whisper 1m

20

0

3

2

10

20

Perceptible change

Threshold of hearing

Barely audible change

1

0

dBFS

dBW

dBSPL

dB

Relative

0dBW=1 watt ref.

Absolute.

0dBSPL=20 micropascals ref.

Absolute.

Listening Perception of loudness Relative

describing very large numbers as much smaller numbers. A logarithmic scale is a good fit with the way we humans perceive sound.

Here’s a quick example: The range of human hearing (change in sound pressure level) is measured in micro-pascals (μP). The threshold of hearing is 20μP, the loudest rock band might be >20 million μP. You can see the range is huge, but described as a decibel it registers as a change of 120dB. Now, the complete range of human hearing is more like two billion μP! ABSOLUTE & RELATIVE DECIBELS

As a general rule, adding a suffix to a decibel makes the unit absolute. Absolute decibels represent comparative change from a known reference. For example 0dBu is defined as 0.775 volts RMS, 0dBV is defined as one volt RMS, 0dbW is defined as one watt, 0dBSPL is defined as 20 micro-pascals. So, for instance, 22dBu is a change in voltage of 9.75V from the reference level of .775V. You’ll see absolute decibels used in technical specifications for mixing consoles, microphones, compressors, etc.

Relative decibels, on the other hand, measure the proportional change between two values. For example, the proportional change from 5V to 8V is 4dB.

Now I promised not to cite maths so you’ll have to take my word for the above solutions. The important thing to grasp is the difference between absolute and relative. There are a huge number of suffixes in the audio industry and beyond audio the decibel is used in industries as diverse as telecommunications, satellite and antennae design, RF design, radar and microwave technology to name a few. Each has its own set of useful reference suffixes.

So which are pertinent to the modern audio practitioner? It’s my view that there are four key decibel descriptors that are important to understand: • dBu (Un-terminated) because it’s the standard quantifier for pro audio: +4dBu=0VU • dBV (to a lesser degree) because it’s still used in many pro audio specs: as a ‘rule of thumb’ dBV = dBu + 2.2 • dBFS (Full Scale) because it’s the metering system almost everyone who is recording digitally will be utilising.

• dBW (Watts) because it’s the standard for pro audio amplifier power description.

• dBSPL (Sound Pressure Level) because it’s the accepted standard descriptor for loudness.

A quick caveat regarding dBFS: Full scale is,

in fact, a relative measure. Think about it. There’s no reference quantifier because full scale refers to 0dB being the point at which your audio will go into clip… but that point is dependent upon your bit depth. Bit depth has no relationship to audio resolution — it defines the dynamic range of your digital audio. At 16 bits, your dynamic range is ostensibly 96dB; at 24 bits, it’s 144dB. It’s a little more complex than this, but bottom line, the point at which you ‘run out of bits’ is variable.

Much has been skated across here. If you’re searching for a more comprehensive overview of decibels that includes the logarithmic equations, relationship to gain structure, dynamic range and headroom I wrote a much more comprehensive gain setting tutorial in Issue 47, which you can dig up on www.audiotechnology.com.au by searching ‘Gain Structure’.

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STUDIO FOCUS:

90° STUDIO TRAINING FACILITY Any musician or studio-type in the Hunter Valley will know about Musos Corner in Newcastle. It’s one of the highest profile and largest music retailers around. Seeing how musicians are now expected to have production chops, Musos Corner has responded: “Nowadays, anyone can record and produce their own music. You can even record with your smartphone or tablet”, says Allon Silove. “But professional studios still exist and with good reason. It takes a lot of time and effort, knowledge and experience to become a good sound engineer. Even longer to become a great producer. We are here to try and help bridge that gap between having the technology and actually knowing how to use it!”

Allon Silove is talking about the new 90º Studio Training Facility (of which he is the Director), a multi-room facility where it’s possible to get hands-on with the latest gear (from names such as SSL, Neve, Universal Audio, AKG, Neumann, API, Empirical Labs and Yamaha) and get some AT 42

studio training while you’re at it. There is also a fully functioning Recording Studio Showroom where you can listen to and compare studio equipment brands just as you would a guitar or amplifier. This is a highly-equipped recording studio that delivers commercial audio training for all levels of skill and interest.

True to its name, 90°’s two main control rooms are square, which required plenty of acoustic work.

Amber Technology’s Daniel Sievert was called in to consult and a brought a swag of Primacoustic solutions with him.

Sievert produced a 3D room model for the acoustic treatment plan: “The design of all the rooms had to incorporate a substantial amount of infrastructure including cabling, TV/computer screens, air-conditioning, ceiling/wall mounts, windows and doors. I offered suggestions of treatment that would work harmoniously with the planned infrastructure while being flexible enough to re-arrange if layouts and infrastructure changed during the setup process.”

Silove reports on the result of the treatment plan, stating that the tracking room: “is now a relatively ‘flat’ acoustic space, able to facilitate an entire band-at-once session while still maintaining a comfortable SPL for working and performing.”

Sievert also offers his opinion on the spaces after treatment: “It was definitely clarity of the sound in the rooms that I was most impressed with. The rooms still have a pleasant ‘space’ to their sound but the treatment seems to be removing any ring or echo.”

Musos Corner has a history of providing education — conducting professional development programs for secondary teachers — so the 90° move was a natural progression. Allon Silove explains some more: “This is a facility offering full-term and short courses catering to everyone interested in music making and music production; at all levels. You might be a beginner wanting to learn how to record or mix your own music, or perhaps you want to develop your knowledge and practice of audio engineering and producing.

Maybe you just want to know how to use the technology you already have. Our courses are designed to be highly practical, giving you the knowledge and experience you need to advance. We are even planning Master Classes for professionals, with some of the biggest names in music production in the Australian Music Industry.

Music Teachers are also welcome to expand their Professional Development with one of our approved education courses.”

90º Studio Training Facility: 0421 490473 or 90deg@musoscorner.com.au

AT 43

STUDIO FOCUS:

NYLON STUDIOS Timing is everything in music, but for the advertising composition and sound design gurus at Nylon Studios, it’s mission critical.

“Someone in New York might ring up at 2am in the morning needing a guitar part re-recorded, and they need it in 10 minutes,” says Music Director Mark Beckhaus. It’s a glimpse into the go-fast or go-home world of delivering for high-profile clients like Sony, Nike and Toyota. A world that has undergone drastic changes since Creative Director Simon Lister set up the business in Sydney 15 years ago. Both Lister and Beckhaus came from old-school analogue backgrounds, Lister worked at Marmalade in New Zealand and Songzu in Sydney, before getting the backing to start Nylon.

But when the simplicity of doing everything in-the-box became obvious, Beckhaus reckons he was one of the first to declare there was no longer a need for recording studios. Then in the GFC, businesses similar to Nylon started shedding inhouse staff like a dog its winter coat. Nylon stuck AT 44

to its guns though and decided to keep its in-house creative team. Currently, there are three full-time composers and three full-time sound designers in Sydney, and four full-time composers and one full-time sound designer in the five year-old New York office.

Having held onto this ‘old school’ in-house model, Nylon also recently decided to return to its analogue roots, renovating one of its ‘guy with a computer’ studio spaces into a new recording studio space and live room. It not only means the composers and sound designers are once again surrounded by warm and toasty vintage gear, but it’s a point of difference in what has grown to be a very vanilla market.

“We went through the whole period of just using computers, plug-ins and virtual instruments.

Now everyone’s doing it, and it’s becoming very generic,” said Beckhaus. “In our business [a recording space] is kind of unusual, because it’s very deadline-driven and price-focused. So people

aren’t taking the time to record drums and things like that. We’re hiring as many players as we can and sometimes using people from bands as opposed to session musicians to get back to what was interesting to us in the first place.”

The new studio space has some select pieces of outboard gear, ADAM monitoring, and an Allen & Heath GSR24M 24-channel analogue desk with motorised faders and DAW control. While Nylon still uses its fair share of available digital tools, the space affords in-house composers the ability to incorporate real instruments into their work like live drums and overdubbing real strings over virtual instruments.

Nylon staffers also get a lot of use out of the space for their own musical projects, and Nylon also takes on the odd film ‘for the love of it’. The latest one they’re working on is a Jeremy Simsdirected effort featuring Jacki Weaver. Nylon also uses the space to put on band nights where Nylon showcases new acts to the advertising industry.

“We’re a bit of a conduit for record companies in that way,” said Beckhaus.

Nylon isn’t at the cutting edge of technology — preferring simple gear that works — and haven’t upgraded their ProTools TDM rigs yet, though it is imminent. With four studios running in Sydney, every rig needs to be mirrored, plug-in authorisations and all, so any producer can work from anywhere in the Nylon system. It means upgrading is a serious concern.

With the need to access any session from any studio, all of Nylon’s files are stored on Synology RAID NAS systems, addressable over CAT5. “We’ve got every job we’ve done since 2002 on the system,” said Lister. “We’ve got a Synology system in the building, and another offsite in a building called the Coach House in the carpark. It’s a complete backup, so we have double of everything. I’ve had five or six rooms running off the same RAID system, and I’ve had a feature loaded up in two rooms bouncing out 120 tracks at the same time without a problem. It basically loads everything into your RAM on the computer. We’ve had no hassles with it. We’ve also got it in our New York office, so we can jump onto that computer, pull a session and it comes straight down to our system here. We were thinking of having both systems talk to each other, but there wasn’t enough happening to warrant that.”

The new studio has reinvigorated the Nylon staff.

Even Blair Joscelyne, one of the Nylon composers and AT writer — who could happily toil away on a 002 with an SM57 if that was all he had — loves the new setup. If nothing else, it’s a great place for clients to hang out and catch a bit of Nylon’s vision.

And that’s worth its weight.

AT 45

REGULARS

PC Audio The latest buzz words in the PC world are Intel's Wellsberg and Haswell-E. Are you up to speed yet, or are they irrelevant to your musical world?

Column: Martin Walker

Well, after last issue's brief foray into nostalgic audio hardware, let's fix our sights firmly on the future and take a look at what PC hardware is on the horizon, and how it may affect the musician. Intel continues to stay ahead of the processor game, and as we've come to expect from this manufacturer, its new generation of processors also means a new motherboard chipset and hence a new motherboard — in other words, to take advantage of all the new goodies I'm about to discuss will mean a replacement PC, rather than an upgraded component or two. However, with promises of performance hikes of '50% or more', there is bound to be some cheering from multimedia types who still long for yet more processing power, and particularly from those who work with bandwidth-gobbling video as well as audio streams.

Right, down to business. Intel's new X99 chipset, going by the family name 'Wellsburg', will be appearing shortly on motherboards from the likes of Intel themselves (and a handful of other manufacturers including Asus, EVGA, and Gigabyte), and will partner Intel's new Haswell-E (for Enthusiast) processor family. Up until a few years ago, processor manufacturers managed to routinely achieve 50% improvements, or even double CPU clock speeds with each new processor family, but once we reached the dizzy heights of 4GHz it proved difficult to continue this trend due to increased system power consumption and cooling issues. So, for the last few years, computing power has been increased not by upward increases in clock speed, but sideways, by running multiple slower processors in parallel.

Haswell-E continues this trend, offering models with six or eight physical cores and clock speeds between 3 and 3.7GHz, each core having the usual Hyperthreading feature that gives it two logical cores. So, this time round your choice of processors will be offering either 12 or 16 logical cores, as well as up to 20MB of on-board cache RAM. Ironically, the musician/multimedia artist may benefit from such an architecture more than many mainstream users, as we tend to run lots of individual plug-ins/ softsynths that developers have cleverly managed to share neatly among the cores. It did take some AT 46

DAW developers considerable time to optimise their core-sharing algorithms as we moved from dual to quad-core (eight logical processor) CPUs, but having mastered eight it should hopefully be far easier to extend the existing code for 12 or 16 logical cores.

MAKING NEW MEMORIES

The new Haswell-E/Wellsburg family also includes quad-channel memory controllers supporting new DDR4 RAM, which itself offers a combination of faster clock speeds and larger module sizes — it will be initially available in 4GB and 8GB modules, with a promise of 16GB variants in the future, and with top-end motherboards having sufficient memory slots to support up to 128GB for those that need it. DDR4 RAM will be capable of running twice as fast as its DDR3 predecessor, but will operate at a significantly lower 1.2V compared with 1.5V or 1.65V, thus reducing power consumption.

Meanwhile, the X99 motherboard chipset will support more USB3 ports, and a new SATA Express format that offers 67% faster throughput than SATA 3.0, to suit the new generation of Solid State Drives with transfer rates of up to 10Gbps.

Before your eyes glaze over with processor lust and numerical overload, let me bring you back down to earth by pointing out that as usual with the latest cutting-edge technological innovations, they will initially cost a lot, until economies of scale inevitably bring prices down. At launch, the new motherboards and processors may attract prices of perhaps US$1000 each, while the DDR4 RAM will be at least 50% more than its DDR3 predecessor.

WELLSBURG WORTH IT?

So what does all this mean for the musician? Well, your mileage may vary, but for many of us it will be business as usual. My PC built in October 2012 is based around a quad-core Ivy Bridge 3.5GHz processor over-clocked at 4.4GHz, and has yet to break into a sweat with any of my projects — like so many other musicians I already have more CPU grunt than I routinely use, and offhand I can't think of any computing situation thus far that has required 100% of its processing power. Those who routinely run over 100 simultaneous audio tracks

won't need Haswell-E/Wellsburg either, as they can already do so quite easily with physical hard drives — really, it's only the combined consumption of multiple plug-ins and softsynths that puts a strain on our CPUs. So, I think the musicians who could benefit from the new Haswell-E/Wellsburg products will fall into three main categories. The first are those who run lots of high-end softsynths, some of which can still only use a single CPU core. Even with a quad-core processor, just one of those softsynths playing 16 or more notes at its 'ultimate' audio quality setting may consume the entire capability of that core.

Now, if you like making synth-based music using half a dozen such softsynths in one track, one of the new Haswell-E processors with six or eight physical cores may be a tempting proposition.

The second category are those exploring the latest virtual console plug-ins that emulate the personality of specific analogue mixing desks, where each track ideally needs its own plug-in instance to achieve the accumulated effect. With complex mixes you may therefore need a hundred or more such plug-ins in place, even before you add a host of other incidental effects on specific channels, as well as several global reverbs. Once again, the parallel processing of the Haswell-E series, along with the faster throughput of DDR4 RAM, is likely to benefit such scenarios.

The third category of musician has always pushed the computing boundaries, and that's the film/TV composer who mocks up entire orchestral scores using sample libraries, where the new DDR4 and SATA Express may together finally enable them to run the entire arrangement on one computer, whereas in the past they have had to split it across multiple machines.

One other aspect has yet to rear its head, and that's the inevitability that software developers with 50% more computing power at their disposal will start creating yet more powerful, more realistic, and more expressive instruments and effects for us. Who knows — we may soon reach the fabled point where it becomes almost impossible to tell whether a composition is created with analogue or digital gear!

REGULARS

Apple Notes Free plug-ins for a rainy day.

Column: Anthony Garvin

I have a love-hate relationship with plug-ins, and I’m probably not alone on this. Reinstalling my software on a new setup recently, I realised just how many plug-ins I had, and consequently, how many I didn’t really know how to use. ‘Jack of all trades, master of none’ comes to mind.

It’s probably not helped by how easy it is to become infatuated with that new plug-in that creates that sound. But really, how many plug-ins are actually essential to creating a good mix? In fact, in my case, having so many I don’t really know how to use is probably hindering my potential for a good mix!

So upon reinstalling, I decided to keep it as minimal as possible — and for the last few months I’ve been working successfully with (almost) one EQ, two or three compressors, one limiter, a couple of reverbs and delays each, and just a few useful metering/utility/‘effect’ plug-ins. Time will tell if this works for me long term, but it has so far.

Now having said all that, we all love free stuff — particularly when it comes to plug-ins! But my approach to assessing the actual value of these has changed, from both the perspective of ‘does it help me?’ and ‘how much of my time does it waste?’, as well as the fact that I don’t really need any other plug-in that imbues my mix with the perfect amount of tube warmth/amp crunch/silky tone/fat compression, free or otherwise. So it’s usually the more interesting offerings that attract my attention.

These days I’ve implemented a screening system for any attractive newcomers. I download the plug-in, but don’t install it, leaving it in a ‘Stuff to try’ folder for the moments where I’m either feeling geeky, or just need to try something different. All the while, I’ve been doing my best to make sure I know how to push the plug-ins I already have in new and interesting ways.

Today I’m feeling geeky, so I thought I’d share a couple of free plug-ins, both new and old, that may help you find quicker solutions for ‘width’ and ‘thickness’.

Acon Digital Multiply AU, AAX and VST (PC too) www.acondigital.com

Brainworx bx_solo AU, AAX, RTAS and VST (PC too) www.plug-in-alliance.com Good for: Monitoring (and simple processing) of individual, L, R, Mid and Side signals.

Quick Tip: For width control and enhancement play with ‘Stereo Width’ to shrink or widen your image (but check your correlation with the latter).

The Rundown: The wonderful thing about this plug-in is just how simple it is to use, whilst being incredibly useful. By being able to easily ‘solo’ not only the left or right channels of a signal, but also the Mid and Side channels, I can quickly assess its width and stereo balance. This is particularly useful on super-wide synth basses that are so common at the moment — listening for low-end in the side channel is an instant check for muddy/ tightness issues. bx_solo can also expand or shrink the stereo image from a mono sum to a ‘400%’ wider version of itself, which is kind of like Live’s ‘width’ control in Utility, or Logic’s ‘Spread’ in Direction mixer, but with some extra trickery going on to enhance the side channel. Word of warning though in Brainworx’s own words: “setting the stereo width control too high can cause serious phase problems!

Always watch the correlation meter.”

Good for: ‘Thickness’, ‘Width’ and kooky echoes.

Quick Tip: Filter out the low end with the equaliser, turn up the voice count and crank the stereo spread. Then wind in the effect level to get an interesting thickening/widening effect.

The Rundown: A relatively unknown developer (well, to me anyway), Acon Digital recently released this free plug-in, and it ended up winning KVR Audio’s Developer Contest of 2014. Touted as a ‘chorus effect with a unique twist’, it’s the unique twists that I have found most pleasing to my ears.

On face value, the plug-in is a chorus effect, with simple frequency modulation and depth controls to honour that claim. It also has amplitude modulation (i.e. tremolo) control and, rather uniquely, a very useable EQ for filtering lows and highs, as well as tweaking the middle frequencies.

This is particularly useful for keeping the low-end tight, whilst nicely modulating (and enhancing) the mids and highs.

But digging deeper, the plug-in becomes very useful for thickening and widening elements of a mix. The Voice Count control can help to subtly thicken up a sound with a minimal amount of obvious chorus-ing like other effects. The Stereo Spread parameter helps to push those voices left and right, to achieve a sense of width that is not obscene when it is in, but sorely missed when it is out.

Multiply also has a pre-delay that can be wound up to 500ms (or one beat at 120BPM), meaning it can be used as an interesting modulated echo/delay.

Unfortunately it doesn’t have internal feedback, however placing this effect on a send, muting the dry level, and sending it back to itself (for feedback) will allow for some very interesting fluttery/ crunchy/thick/swirly echo effects indeed!

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QUICK MIX

The

with

Gavin Tempany Interview: Neil Gray

Who are you currently touring with/mixing?

Tame Impala.

What are some other bands that you have worked with?

Missy Higgins, Eskimo Joe, Orchestral Manoeuvres in the Dark, Little Birdy, Evermore, Dropkick Murphies, These New Puritans.

How long have you been doing live sound and how did you get started?

Since about 1993, so I guess that makes it over 20 years. I was that guy in the band who did the recordings and was interested in the tech stuff. Then I did the WAAPA course and fell into live sound in-between recording bands. I have done a lot of theatre sound as well, which is really good for getting your mixing chops up.

What is your favourite console and why?

I have been using the Midas digital consoles of late. I really like the Pro2C. It’s compact and has 64 inputs to mix. Not to mention it sounds like the Midas XL8, which was the first digital console to really sound good. But to be honest I use all consoles. It’s more about the driver than it is about the console.

Favourite microphone or any other piece of kit?

This is tricky. I have so many gadgets it’s hard to single one out! I am very keen to see how the new Soundcraft/Universal Audio plug-in box goes. I tried it a while back and that thing sounds ridiculously good.

Most memorable gig or career highlight?

Just recently I mixed a charity event at Kensington Palace in London.

The last act of the night was Bon Jovi. He called out to Prince William to come on stage with Taylor Swift and sing a karaoke version of Living On A Prayer. Highlight was Taylor Swift and Price William highfiving after singing the first chorus! Classic. Not your everyday gig!

Any tips/words of wisdom for someone starting out?

Learn how to do all the aspects of live sound. Flying PAs, tuning rooms, tuning wedges, how to pack a truck well, how to work in a team. Then when you start mixing bands you will understand the signal flow better and this directly translates to your mixes and lets you solve problems quicker.

Describe your mixing setup now, compared to what it was in 1998.

On this tour I have a little Pro2C and an external computer inserting some effects. To be able to do what I am doing now at the same quality in 1998 would have meant racks of effects and a massive mixing console. Having said that, 15 years ago, I was not touring a console but used to carry small racks of outboard that I could fly on planes — now it’s USB sticks and iLoks.

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What are three mixing techniques you regularly employ that you’ve learnt in the last 15 years?

Although not a mixing technique, I would hope I’ve learned to hear problems and fix them quicker. Everyone is mad about parallel compression. It works, but it’s not the golden goose. Watch out for latency in digital consoles. I think really it’s still about balance in a mix. I just try and get it so you can hear everything! Forget the rock star roadie. Isn’t that what we’re there to do?

In the last 15 years, what are three pieces of gear or features that have come out and been game changers for you?

There are really quite a few. Take your pick: Lake Processors — What a genius interface and sound quality, one of Bruce Jackson’s legacies.

Digital consoles — Show files mean you could finally travel a show without having to travel the weight of a console. It has brought the standard of audio up remarkably on smaller tours that cannot afford to have a truck due to everyone having a rough starting point for their shows — not to be underestimated at a festival. Only now you need a show file for every console because a 15-minute changeover might not be enough time to tick every option!

Switch Mode power supply power amplifiers — This has got to be one of the main changes over the last 15 years. Initially distrusted for their (wrongly) perceived lack of low end, they are now used on all parts of the system due to their power to weight ratio and incredible transient response. So much more power than 15 years ago.

Virtual Soundcheck — Super handy for rehearsing bits of a mix up and generally prepping for a tour.

Line Array — This has been the buzz for the last 15 years. It certainly gives real estate back to video to put up screens! To be honest nearly any PA system that has been designed in the last 15 years is useable out of the box and tonally pretty close. Takes the fun out of it!

How have your working methods changed over the last 15 years?

Not really sure I am doing anything super different. Although I feel I am better now at what I do because maybe I do less!

AT 49

REVIEW

SE ELECTRONICS RN17 Small Diaphragm Condenser Microphones Rupert Neve leaves his signature on these sE microphones in more ways than one.

Review: Brad Watts

NEED TO KNOW

There’d hardly be a single audio product manufacturer that wouldn’t knock back a collaboration with Sir Rupert Neve. Yes, I know Mr Neve hasn’t been knighted, but if Sir George Martin can be bestowed such royal privilege then Rupert certainly deserves the monarchial nod — the man is responsible for the recording industry’s sweetest tracking consoles dating back to the mid-1960s after all. Even Sir George Martin himself relied on Rupert’s Focusrite entity to create behemoth consoles for his Air Studios. The list of Rupert-driven manufacturing

PRICE $1499.99 each $2999.99/stereo pair Capsules start at $449.99 each

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CONTACT Sound & Music: (03) 9555 8081 or info@sound-music.com

entities and collaborations is testament to the man’s insurmountable legacy: Neve, AMS Neve, Focusrite, Amek, Legendary Audio, Taylor Guitars, and most recently, sE Electronics with its ‘Rupert Neve Signature Series’ microphones.

Which brings me to the matched stereo pair of small diaphragm condenser mics I have perched in the studio today; the sE Electronics RN17.

From the second I’d set eyes on this package I was unashamedly impressed. The microphone pair come in a superbly crafted timber case, which is in turn enclosed in a very pucker

PROS Superb SDC Detailed, neutral, ‘musical’ Pile on the EQ Superb construction

aluminium flight case — and not the standard Jaycar, made-in-a-hurry case either. This case has fancy catches and a big chunky handle too.

In other words, this is a top-shelf product, with no economy spared. Upon opening said spiffy flight case, eyebrows were raised as I checked out the shiny metal screw-top canisters with tell-tale polar-pattern symbols embossed on top — ten of them in all, covering five polar-patterns. This was unfolding to be a splendid box of microphonic fabulousness.

Unfortunately I found these canisters empty

CONS Tranny bulge makes some stereo positions awkward Price of family set gets up there

SUMMARY The sE/Rupert Neve collaboration yields what you would expect — transformers, and classy sound. Stereo positioning can be fiddly with the tranny bulge, but the full-bodied small diaphragm sound is worth any hassles.

as the mics ship with cardioid capsules only, with further supercardioid, omni, hypercardioid and figure-of-8 capsules available optionally. Despite my disappointment with the lack of polar-pattern options being supplied for review, I must say the canisters are a lovely touch — guaranteed to keep your additional capsules away from moisture and other travesties that may occur during use.

But again, all this goes to show the no holds barred approach to these mics, even the serial numbers are indicative of a matched stereo pair, (i.e. xxxxA and xxxxB) — and I’ve not even mentioned the brilliant shock mounts, which I’ll get to a little later.

SMALL DIAPHRAGM, HIGH BAR

The RN17 is essentially a small diaphragm condenser microphone, and consequently is aimed at all the tasks you’d associate a SDC mic with.

Stereo recording situations, drum overheads, basically anything where you’re aiming to capture accuracy and fine transients, be that acoustic guitar (or similar) or even hi-hat and ride cymbals. But, as always, rules are meant to be broken and a SDC mic could well be the ideal option for a particular vocalist. Mileages vary.

So why is the RN17 any different to any other SDC mic? Well here’s the big calling card with the RN17. Unlike most SDC mics, the RN17 incorporates a large hand-wound, “ultra-high performance” transformer — hence the large protrusion at the connection end of the mic.

I’d highly suspect this portion of the design is courtesy of Mr Neve. Why use a transformer?

There’s a number of advantages to transformerbased designs, firstly that of a transformer’s ability to reduce noise, or increase the mic’s signal-tonoise ratio. Without going into the maths involved, a transformer should be matched to the preamp it’s married to, with an appropriate number of ‘windings’ to achieve the best balance. But aside from this technical advantage, transformers lend a particular ‘character’ to a signal path’s audio quality. The interaction of level and impedance result in a desirable sonic response. So with the RN17, sE Electronics claim to have created the world’s first transformer-based small diaphragm condenser microphone.

But enough of the chit-chat. The question is; how do these mics stack up. Well let me start with the fact these mics are a pleasure to set up, courtesy of the rather brilliant (and aforementioned) shock mounts. The mounts are a kind-of dual rubber-banded mounting system.

With adjustment for distance between the two spiderweb sections. They’re like no shock-mount

I’ve seen before and frankly, I’d like a pair. I did come into a little bit of strife with the design, however. Ideally it’d be great to mount the RN17s into their shockmounts with the ‘transformer bulge’ sitting between the two support frames of the shockmount — the mics physically ‘sit’ better in this position. Unfortunately with the overall length of the mics and the width of the supplied (and very nice I might add) stereo bar, it’s virtually impossible to set the pair into a standard X-Y stereo configuration. To achieve this configuration accurately the RN17s ‘transformer bulge’ needs to sit on the outside of the rear support frame.

If the stereo bar was slightly wider this could be alleviated and the mics mounted with the trannybulge in the more stable centre position. The bulge also makes ORTF configurations fairly fiddly, risers would also be a helpful stereo bar update.

CUTTING THE UPPER CRUST

On first listen to the RN17 I was immediately convinced I had a pedigree microphone in my hands. My initial, somewhat ad-hoc, test of a mic is to listen to my own voice via the mic and headphones. It allows me to get the mic into ‘announcer gain mode’ and get a handle on how delicately the mic will perform with a source I’m intimately attuned to; my own voice. In this scenario these mics proved impressive. Very detailed and bolding wearing their transformer heart on their sonic sleeve. It was quite a revelation hearing the detail of a SDC combined with a transformer. Very nice indeed.

For a more realistic comparison I set the RN17s in X-Y fashion along with a pair of rather modified Oktava MK012s, and a pair of recently acquired Neumann KM184s. To be honest, after this first audition, I’d rather the RN17s over the KM184s. But my pockets aren’t that deep and the Neumanns were too good a price to pass up.

Anyway, all three pairs were set in stereo X-Y and pointed toward my favourite spot in the room for acoustic guitar. Out came my trusty 1970s Sigma, six tracks were bumped into record, and some playing was had. Remember I’d only had access to the standard cardioid capsules and consequently used the same style capsules on the Oktavas, the KM184s are of course cardioid.

Upon playback, the most marked difference was the bass response attained via the RN17s. I’d almost reached for the high-pass filter when I realised this would require a different capsule. That said, the lows didn’t detract from or swamp any top end capture — the RN17s sounded smooth and remarkably realistic. Compared with the other contenders these were by far the superior choice, with the Oktavas sounding like their usual sparkly/pretty selves, and the Neumanns sounding almost bland by comparison. (I often have this feeling about KM184s but I know they work for so many sources when adding them to a mix).

Where things got really interesting

is when applying EQ to the three recordings. The Oktavas don’t really appreciate too much tweaking in this department — any extreme sculpting is usually met with resonant squawks and honking, and typically renders the recording difficult to sit into a mix. If it’s not right flat then it’ll never work with those mics. I find a similar issue with the KM184s, although you can achieve far more using EQ than with the Russian cheapies. Once you start EQing the RN17 recordings you can hear just how sublime these mics are. You can carve into the recording like there’s no tomorrow without the recording becoming brittle and unnatural. It’s this characteristic that makes the RN17 such a beautiful microphone. Fine detail and pure, neutral capture and the scope to be able to alter that signal without nasty degradation. Spec-wise the RN17 is closely matched with the KM184, however it will absorb a further 8dB of level before distortion at a whopping 150dB SPL. So would I own the RN17s? Well, I think the answer to this is blatantly obvious; a resounding, “Hell yeah!” But alas, these are expensive pieces of audio capture equipment and I can’t justify the cost for myself, especially considering the cost of branching out into further capsules. However, if you’re after nigh-on-perfect small diaphragm condensers, I’d advise making sure you can afford these before auditioning a pair. You’ll simply not want to send them back.

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REVIEW

ANTELOPE AUDIO ZEN STUDIO USB Audio Interface Antelope Audio has released the most flexible USB audio interface on the market, but can you get a handle on it?

Review: Mark Davie

ON THE BACK

NEED TO KNOW

As well as analogue inputs on combo jacks, you also have eight analogue line I/O on D-Sub, two analogue insert points, a main monitor output pair, word clock I/O, S/PDIF I/O, and 16 ADAT I/O on four ports around the side.

PRICE $3199 CONTACT Soundtown: (08) 9242 8055 or www.soundtown.com.au

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PROS Loads of I/O Clean preamps, great clocking & conversion Stable USB drivers Software control panel very flexible

CONS No MIDI Lack of dedicated hardware monitor controls Some may want to rackmount it

SUMMARY Antelope’s Zen Studio gives you the clocking and conversion technology of the Orion32 in a portable USB audio interface and adds 12 preamps. The only thing you’ve got to mull over is whether you’d rather rack ears or the most flexible interface on the market.

When images first surfaced of Antelope Audio’s Zen Studio, the most talked about feature was its handle. The single, fire engine-red piece of bent metal really caused quite a stir. A lot of people thought designing an interface that looked like it would fit into a rack means it should actually fit into a rack (sure enough, I’m reliably informed that Antelope is developing a rackmount kit that leaves 1U space above to ensure the Zen Studio stays cool). As useful and as intriguing as the handle is and was, it’s still a ballsy move on the part of Antelope. Yes, portability is a huge asset, but there are many people who would prefer to squirrel this baby away.

But here it is, and the Zen Studio hasn’t changed from those initial shots one bit, handle and all.

The other main arguments for keeping the Zen sans rack ears are related to the device’s high I/O count. Firstly, it gets a little warm, so Antelope has designed ventilation into the top and bottom of the unit, with some elevating rubber feet to let it breathe. Secondly, the four ADAT expansion ports are on the short end opposite the handle, so you’d have to incorporate some right angle adaptors if you wanted to use them in a rack.

In summary: the Zen is designed to cater for the kind of recording a lot more people are doing these days. Namely, capturing ‘out in the field’ — practise room, church hall, club, whatever — and heading home to mix.

and outputs. So what’s the trick?

The Zen Studio uses the same Burr-Brown PGA2500 preamp chip as Universal Audio’s Apollo, Apogee Duet, Prismsound Orpheus, MOTU 828mkIII, RME Fireface 400, and plenty more.

And like the rest, the digitally-controlled preamp has a flat, even response, with a gain range of 55dB from 10dB to 65dB. Obviously, there’s variation in implementation, but the basic gist remains the same, and Antelope definitely hasn’t throttled its performance in the Zen. I don’t mind these pres at all, using them on the Apollo Twin was fine — good clean gain — likewise here. Obviously you don’t get the special sauce of Apollo’s Unison preamp modelling on the way in. Though interestingly, because both units use the same preamp, when using the Zen you can insert the Unison models in your DAW and achieve some similar harmonic effects, albeit without the same gain-staging.

Fitting so much into an almost 1U space requires a bit of digital fandangery and a reduction in physical controls. In the Zen’s case, there’s one major dial and three buttons on the whole device, control is essentially deferred to the software Control Panel. Most users won’t be perturbed by the inability to adjust gains on the front panel, but you might find these soft controls takes away one of the few tactile operations left.

MEDITATE ON THIS

The digital preamp control does allow for some nice features. For instance, the gain tracking is very solid, and you can stereo link consecutive channels; fixed as odd numbers on left, even numbers on right, so you’ll have to plan your channel assignments early.

There are a few options that are about to land that are worth noting. Along with the 55dB of gain, the PGA2500 also has a unity gain feature (about to be implemented), while a pad and output trim will also be on the final incarnation. Antelope has been working on techniques to switch to digital gain at lower levels to give users control down low, then kick back into analogue gain for the higher levels — hence this feature not being on the review unit.

The idea is the Zen Studio packs in everything you need to record most projects, and the ability to expand when it doesn’t. There are a total of 38 possible inputs, and 32 possible outputs, and all are simultaneously available. Available in the sense you can mix them using the Zen’s software control panel and onboard DSP. When it comes to recording, Antelope’s USB chip provides 24 I/O, without sacrificing any channels to go higher in sample rate — a common issue with other interfaces — all the way up to 24-bit/192k.

The USB drivers and chips are exactly the same ones Antelope uses in the Orion32, which can handle 32 I/O over USB at 192k, so if anything, the Zen should prove more stable.

On my 2.6GHz i7 Macbook Pro it hasn’t skipped a beat. Set up was painless, and there haven’t been any idiosyncratic annoyances like having to power up my computer and devices in a certain order, which can be painful sometimes. And letting it fall into Sleep mode and jolting it awake made no difference to the Zen Studio — it just meditates while waiting for your next move.

PREAMP CAMP

There are a total of 20 analogue inputs on the Zen Studio. There are 12 combo jacks — four on the front that have a three-way function: DI, mic or line; and eight on the rear that are switchable between mic or line. The other eight analogue line inputs are available on a D-Sub connector. 12 preamps will go a long way for a lot of situations, but it’s also a lot to pack into an almost singleheight rack unit along with all those other inputs

DIGITAL DIVINATIONS

there’s a stupid amount of I/O for any 1U-ish audio interface, let alone a portable one

CONVERSION FROM THE STARS

The Zen Studio has the exact same converters, clock and USB chips as the much-heralded Orion32, the only thing it misses out on is a 10M atomic clock input. The 10M would be complete overkill for the Zen anyway, and besides it has one of the best internal clocks going.

All the standard clock sources are available: internal Oven clock, ADAT, S/PDIF and word clock. Also, when you start up a DAW, the clock source and sample rate greys out and locks to USB. Essentially, this still clocks to the internal ovencontrolled clock, it just means all the sample rate control is handed over to your DAW so you can’t set conflicting rates in the Control Panel. For more insight into the clocking technology and quality of conversion, search for Andrew Bencina’s review of the Orion32 on www.audiotechnology.com.au.

Andrew describes the Orion32 as a ‘fine AD/DA AT 53

1

2

3

1 The software Control Panel is a real pleasure to use.

The Routing tab is split into ‘From’ and ‘To’ sections where you can route any input to any output or mix, with colour coding to help you keep track of what’s going where — deceptively powerful.

2 The Meters tab allows you to meter all your inputs and outputs by selecting them from a drop-down menu. And whatever group you select is reflected on the hardware unit’s meters. There’s provision for up to 32 meters on any one screen.

3 You can use the mix engine to create four ultra-low latency, 32-channel monitor mixes, with up to 16 freely routable channel strips that include a five-band EQ and compressor section. The negligible latency of the monitor mixers are really handy for tracking.

converter with unique characteristics’ and would extend that description to the Zen. In use, the Zen Studio sounded great. The outputs are really clear, there was no discernible noise introduced into my monitoring system at any level. The same can’t be said of a lot of audio interfaces. The dual headphone amps are also clear and don’t go exceedingly loud, which is a good thing for protecting your ears.

One thing I would have liked to see is separate knobs for headphone levels. There are two ways of accessing them, one is by clicking through outputs on the main dial till you get the one you want to adjust, or doing it on the software Control Panel.

But I like having quick hardware control over the levels feeding my ears, especially if a computer freaks out and starts trying to deafen me. Leaving the main dial for monitor level and adding two for headphones would have been useful. Thankfully, when you press and hold the volume knob you mute the output you are currently controlling, which is a useful feature.

On the input side, while the 12 combo inputs use a differential amp to handle the varying levels presented to it, the D-Sub line inputs use an alternate amp feeding the AD conversion stage.

You also get more control over line levels on the combo inputs, with a range of -6 to +20dB, while the D-Sub inputs range from -6 to +12dB. Whether line inputs do or don’t bypass onboard preamps is a big concern for some. But in practise, there didn’t seem to be any difference between using a line input on either the combo or D-Sub inputs. When feeding the same signal through each, they almost completely nulled, and what was left seemed like the result of a minute gain matching issue, not any tonal difference.

tab is split into ‘From’ and ‘To’ sections where you can route any input to any output or mix, with colour coding to help you keep track of what’s going where. Once I got the hang of dragging around single channels or groups, I felt it was actually much easier to manage than a typical matrix, which would have to be incredibly wide to incorporate the same flexibility.

Using this method you can easily see what is feeding your monitoring systems, and each of your four onboard mixes; what you’re routing through the onboard DSP effects; which channels or mixes you’re recording; and what’s feeding your ADAT outputs.

As I was saying earlier, you can record up to 24 channels at a time. But if you were doing a live recording and needed more inputs in a pinch, you could premix some tracks together in the internal mixer and make use of all 38 inputs. Alternatively, you can use the mix engine to create four ultralow latency, 32-channel monitor mixes, with up to 16 freely routable channel strips that include a five-band EQ and compressor section. The negligible latency of the monitor mixers are really handy for tracking. Because when using ProTools, I was getting a consistent round trip latency of 5.4ms — plus double whatever buffer I was using at the time — so reducing the latency was necessary for accurate monitoring. The onboard DSP effects are quite good, and more than enough for crafting monitor mixes. Some might lament the omission of reverb and delay, but setting up a pre-fade aux in your DAW with the effect on it will work just fine. And adding more effects to its DSP quiver is another area Antelope plans to work on in the future [Antelope, as of going to press, give the reverb a couple of weeks before it’s released — Ed.] .

The Meters tab is likewise comprehensive, you can meter all your inputs and outputs by selecting them from a drop-down menu. And whatever

group you select is reflected on the hardware unit’s meters. There’s provision for up to 32 meters on any one screen, so it would be nice to have logical collections of meters to make use of that space. As it stands, the 12 preamps and eight line inputs are split across two different screens — it would be handy to view them all at once.

One downside is there’s no MIDI on the unit.

This is becoming more of a theme with audio interfaces, which I find a little annoying given my controller keyboard is an old Korg M1. Maybe it’s time for me to move into the future of MIDI over USB.

CONTROL PANEL

The software Control Panel is a real pleasure to use, despite being unfamiliar at first. The Routing AT 54

GETTING AROUND

Oddly enough, the handle is really the Zen Studio’s only potential turnoff. Without it, and with the ADAT ports relocated, I think it could have been more of a no-brainer for anyone on the interface hunt. It’s not that I don’t like the handle, there are plenty of ways I could imagine integrating the device into my setup in a more permanent way, and a mobile interface is always useful. But having the ADAT ports on the end does make them a bit prone to having their caps snapped off when carting it around. Which is the whole idea of the interface, to keep it portable. Keeping with the portability theme, I’d almost rather have the ADAT ports shielded by the handle.

The Zen Studio really is an impressive device though. The clocking and conversion is supreme, the pres are nice and clean, and there’s a stupid amount of I/O for any 1U-ish audio interface, let alone a portable one. For anyone that needs loads of I/O and has an ear for quality conversion, the Zen Studio can put a peaceful end to your search.

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REVIEW

Audient iD22 USB Audio Interface & ASP880 Microphone Preamp ADC Audient’s desktop interface and preamp rack combo is closer to the company’s console heritage than it looks.

Review: Andrew Bencina

IN CONTROL

NEED TO KNOW

Volume controls on the iD22 are made using its large custom-milled aluminium master knob and applied digitally in 1dB increments within the DSP mixer application. Audient Technical Director Tom Waterman expands: “The volume control is scaled such that dynamic range isn’t compromised significantly when at typical output levels and loudspeaker SPL. The benefits to doing it digitally include perfect stereo matching, its output can be assigned wherever you need it, and it’s very cost effective.” Despite the digital control, the pot has end points and isn’t replicated in the software interface, so storing and recalling preferred level settings isn’t available.

PRICE iD22: $999 ASP880: $1899 CONTACT Innovative Music: (03) 9540 0658 or info@innovativemusic.com.au

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PROS Both preamps & conversion sound great iD22 delivers a full-featured console master section Inserts on all input channels ASP880 ADC & preamps capable of independent operation

AS YOU LIKE IT

The iD22’s three user-assignable function buttons provide direct access to many DSP mixer monitor functions, including: Talkback, Mono Sum, Polarity reverse (used in combination with Mono Sum for mid/side monitoring), and an Alternative speaker send with it’s own trim control. These compliment the dedicated Dim and Cut switches located below the master volume control. Additional settings for Mono mode, Cue mix preview, and global mixer presets are presently only available from within the iD Mixer desktop application. Button modifier combinations are strictly reserved for in-house development functions… for now.

CONS Room for refinement in iD22 DSP mixer/matrix iD22 digital outputs absent from DSP mixer Just average low latency performance under load

SUMMARY The iD22 has some fairly serious competition in its class.

Low latency performance and the flexibility and power of its DSP mixer/matrix falls short of the best but its signal path and master section is formidable. The ASP880’s dual digital interface and modular hookup options make it a viable addition to myriad recording setups. Its preamps offer tone and dimension rivalling those many times its price; and while you wont always use all of its features, they’re nice to have.

AUDIENT INSIDER It’s hard to avoid the review clogging stream of specification features and statistics when compiling an overview of any new product. Audient’s Technical Director Tom Waterman takes AT beyond the numbers for a more detailed view of their design philosophy and implementation.

Tom: “The goal for the iD22 interface was simple: make it feel like a ‘mini Audient console experience’ in a small format. Make it affordable, easy to use and most importantly, make it sound great. Steve Flower, myself and David Dearden (the design legend) worked on the backbone for iD22.

“It builds around the Audient mic preamp; a design which has been optimised by David Dearden over the last 15 years, each iteration perfecting the PCB layout, grounding, noise performance and THD optimisation. The mic pre is fundamentally Class A (same as our console), with eight input transistors running as a discrete gain stage with low noise input pair.

These are ‘linearised’ by wrapping the discrete stage up with an IC op-amp second stage in the feedback loop so they provide 60dB of clean and quiet gain.

“The mic pres and analogue stages of iD22 run on ±15VDC (similar to an API console for example) and provide plenty of headroom and professional output drive compared to a lot of 5V USB interfaces. To that end, the power supply design of iD22 is critical, with eight internal voltage rails provided regulated audio rails, AD/DA reference voltage sources and various low voltage rails for digital processing, etc — each section of iD22 is separated at the supply. Thus the audio rails are well isolated from dangerous switching digital power rail noise.

“We benchmarked against the usual mid/high-priced desktop interfaces (some German, some American

The iD22 is the first of a new line of Audient audio interfaces. The English-designed, Chinese-built new kid quietly honours the company’s near 20-year heritage in console design and manufacture. On face value the two-in, six-out interface seems like an understated addition to a sector of the market offering plenty of choice. At $999 its price alone suggests there’s a little more going on. Supporting multiple flavours of Windows and OSX (10.6.8 or later), the USB 2.0 device relies on the included 1.5A 12VDC wall wart for juice. A desktop design, the chassis is built like a tank and favours a layout placing monitor control at the heart of operations.

I.D. THANKS, KID

The top panel provides control over two channels of Audient Class A mic preamp (a +/-15V version of their console pres) with switches for polarity, HPF (-12dB/octave at 100Hz), +48V phantom and a -10dB pad. On the rear of the unit two combo connectors offer both mic and line access to these channels with channel two boasting a 1MegΩ discrete Class A JFET DI. Both analogue

if you can read between the lines) and decided upon a converter set after much testing and evaluation of various AD/DA chips to find the most natural sounding parts with excellent dynamic range specs.

We settled on Burr Brown PCM4220 and PCM1798 (used in some very favourable hi-end mastering converters) and have optimised the ADC and DAC circuits to provide performance and sound quality we feel reflects the Audient sonic ethos (clean, open and sweet) and would stand up well against the competition.

“Clocking is performed locally by a high quality crystal oscillator and we use a two-stage Cirrus DPLL to regulate clocking and make sure the unit has stable stereo image. From there, we have a discrete headphone amp driver — very similar to the classic Neve BA640 op-amp idea; an op-amp with a current-boosting Class AB transistor output stage — and user-assignable functions for monitor control, directly on the hardware.

“We use some very tried and tested analogue building blocks that hark back to David’s heritage in the ’70s and ’80s of the ‘Best of British audio’ era.

But never seen in the public eye is our relentless optimisation stage which can last up to six months, where op-amps, capacitor types, resistor types are all auditioned and measured to find the optimum sound quality and (measured) performance for the target price. Ultimately the circuits are quite simple because they are distilled down and simple often sounds better. Design is all about implementation and not just parts choice.

“With the ASP880, we started with the same classic Audient mic pre, except it runs on ±18VDC just like the consoles; offering 2dB more headroom than

input paths feature balanced insert pairs prior to the ADC; the first time I’ve seen this in such a model. Two stereo output pairs are delivered via TRS and a third is available via the headphone jack. Both analogue paths and conversion is really impressive and bring some classic console character to a format that often sounds relatively bland. Optical I/O expands the iD22’s capabilities, supporting both stereo S/PDIF and ADAT (eight channels at 44.1/48k, four channels at 88.2/96k via SMUX), making it a viable option for both musicians and small recording setups. During testing I did experience some sync problems when clocking to and monitoring from an external ADAT source at sample rates >48k. It proved to be an unreported Windows specific bug but within a few days a new beta firmware version had been provided to rectify the issue.

GET ON BOARD

A 32-bit on-board monitor mixer is accessible via the desktop application incorporating channels for all inputs and the first six software

the iD22. Eight channels of mic preamp in 1RU is definitely a squeeze, but we designed a completely new PSU which is actually lower noise than the old ASP008 supply, has virtually no mains hum component, and allowed us to remove the fan. Even running power from rear to front in a crammed 1RU case can be a bit dangerous from a noise point of view so there is no power switch. Truthfully, there’s no space on the rear or front panel even if we were inclined to compromise on this.

“The key is keeping the PSU noise as far out of the audio bandwidth as possible (in a switch mode type) and from there, critical layout, managing current loops and using appropriate inductive filtering to reduce any output noise is essential. The benefit to regulating current loops and keeping them as tight as possible, in David’s words, is to “separate the dirty current from the clean current” (grounds in particular) so that we can reduce any mains hum and other gremlins to almost nothing compared to a toroidal linear supply. The other benefit is that with no significant magnetic field in the enclosure, Channels 1-8 are pretty even noise-wise, whereas on the original ASP008, you might find that there is more induced hum in Channel 8 nearest the transformer.

“Although specs matter to some degree, and so do all of these design decisions. When it comes to the crunch, whether a product is a killer unit or not, comes from the benefit it brings to either sound and/ or workflow. If your ears say yes and the workflow is transparent so that creativity can happen, then that is most important and a great benchmark for a successful design.”

outputs for the creation of one master and two cue mixes with approximately 1.6ms roundtrip latency. These mixes can then be routed to any of the hardware outputs via a matrix of radio buttons.

The remaining eight software outputs, relating to the digital interface, have been omitted from the mixer and I missed them when tailoring monitor mixes using an extended combination of input, playback and effect return channels. Limited DSP power means the iD22 cannot compete with those offering full channel strips and auxiliary effects within their software mixers while the iD22 matrix currently overlooks loopback, mirrored routing and hardware output calibration options that can be very useful. There is however a full complement of monitor switching options that will suffice for most applications, and configurations can be stored in presets. With no current support for iOS/Android or TouchOSC control, the iD22 isn’t optimised for standalone operation. It will remember your most recent mix when you disconnect the USB but if you need to change any settings, including sample rate, you’ll need to access it via the desktop app. However, with AT 57

3 kΩ HIGH & FALLING

ASP880 mic preamps feature three selectable input impedance options (Hi: 2800Ω, Med: 1200Ω, Lo: 220Ω). As a general rule, to ensure a properly bridged circuit, maximising signal and fidelity, and minimising noise floor, use a setting >10x the output source impedance of the microphone. On this basis, the Hi setting will be adopted by most as their default configuration, while the lower settings deliver creative variations for sculpting of tone, transient shaping and clarity.

Perhaps a higher Hi setting (6kΩ+) would’ve delivered greater compatibility with some high impedance ribbon microphones, for example, but this is a speciality available on few preamps in the market. Quite reasonably the iD22 preamps operate at a fixed 3kΩ impedance.

BETWEEN THE LINES

In a design decision that speaks to the company’s console heritage and sets both of these devices apart from most of the competition, Audient has included insert points (separate send and return connections) for all input channels. While this obviously makes it easy to incorporate your favourite outboard processing in the pre-ADC signal chain, it also allows users to bypass the preamps, via the return inputs, when you’re after a direct and pristine converter input.

the right configuration it can add two standalone mic pres, and six channels of ADAT DAC to any studio; and even moonlight as a powerful ADAT driven monitor control centre. This is valuable multitasking from a portable interface.

The audio performance of the iD22 is worthy of interfaces more than twice its price, but as the first in a new family of products there’s room for the software and interface implementation to grow.

At the lowest buffer settings the interface delivers only average stability when placed under serious processing load but it’s certainly not alone here and is simply less suited to live use. At this point OSX seemed to deliver performance improvements against the Windows installation, which has been a more recent development. In practise, I didn’t have any problems when recording and the low latency DSP mixer meant I could back off the buffers if things got edgy without any impact to the monitoring experience.

GIVE ME EIGHT

While the ASP880 can be connected to the iD22 via ADAT to form a potent studio front end, the AT 58

eight channel preamplifier and AD converter should be very much in demand on its own terms.

It features the same ADC as the iD22 while the preamps have been upgraded to run at the full console spec. This update of Audient’s previous ASP008 packs its preamps, dual ADAT/AES digital interface (eight channels of 24-bit/96k via either) and Word Clock sync into a single rack unit and subsequently should be installed with an empty space above it for heat management. Like the iD22 it includes balanced sends and returns on every channel (via two D-Sub connectors) and Channels 1 and 2 include dedicated instrument DIs. All channels feature phantom power, polarity invert, a variable HPF, variable impedance and the option to bypass the preamp entirely and run a line level signal direct to the ADC. Again Channels 1 and 2 are given the special treatment and add -10dB pads to the equation. Personally, I’d have been willing to trade in impedance switching or the variable HPF for pads on at least another couple of inputs. Give me an output trim pot and I’ll trade-in another couple.

At rough count, there are currently at least

40 interfaces from 13 different manufacturers who provide either ADAT or AES as a digital expansion option on their devices. Combine this with outstanding performance and you’ve really got something. Using Radial’s Jensen transformerisolated microphone splitters, I recorded a broad range of sources through a combination of preamps and DI channels including: the UA 2108, Phoenix DRS1, Daking Mic Pre IV, Quad Eight MM312, and Radial JDV Mk3. The Audient’s were never out of place and in many cases, differentiating between them proved a challenge. In the end, I placed them tonally somewhere in between the 2108 and DRS-1 with a leaning towards the UA. When you consider the standard practice of pairing very neutral preamps with converters, the ASP880 offers something quite different. For under $250 a channel including conversion, they’re also a very viable alternative to those extra 500 series modules you’re lusting after.

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REVIEW

PSI AUDIO A21-M Studio Monitors Pump up your critical listening setup with some PSI monitors.

Review: Mark Woods

NEED TO KNOW

Studio monitors tend to be functional rather than audiophile, and often chosen on price as much as sound quality, but there is a better world out there. One where working with great sound can be a treat, where so tangible and detailed is the sound, mixing becomes a craft and less like blunt force trauma.

The high-end studio monitor market is led by dedicated audio manufacturers, both corporate and boutique, who continue the noble quest for better sound. Boundaries get pushed and gains are made in controlling the myriad parameters affecting the final sound delivered to the listener. PSI Audio is one of those companies, combining sophisticated electronics, strict objective measurements and

PRICE $11,290/pair CONTACT Group Technologies: (03) 9354 9133 or sales@grouptechnologies.com.au

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PROS Accurate sound Distinctive finish High volume ability

critical listening to make a better monitor speaker.

It is the Swiss way.

TIMELESS DESIGN

Based in Switzerland, PSI Audio’s parent company Relec has a decades-long history making speakers, many on an OEM basis, including a line of studio monitors for Studer in the ’90s. Since 2004 they’ve been making a range of studio monitors under the PSI brand, and Group Technologies has recently begun distributing them in Australia. The A21-M sits in the middle of the current seven-speaker range and it’s aimed at commercial studios, serious project studios, broadcast and surround sound applications.

PSI aims high. The speakers are almost completely hand-made, with unique features

CONS None if this is your price range

inside and out. The cabinets are made in SainteCroix, a Swiss mountain-town with a long history of creating finely detailed music boxes, automatons, and ornate gramophones. While not gilt-edged, I daresay that heritage helped influence the distinctive deep maroon, slightly sparkly finish on the cabinets in the range. They look great, but you can get black if the reddish tone clashes with the décor.

The A21-M is a front-ported, two-way design with the tweeter set well back inside a circular waveguide. A recessed light sits beside the tweeter to indicate power on, and flashes to let you know you’ve found full level. The eight-inch woofer is fully exposed, screw-heads and all. At first I thought it looked a bit unfinished compared to the rest of the cabinet but it grew on me. An optional

SUMMARY These Swiss monitors are incredibly detailed, flat, and loud, without being clinically cold. About as good as you can get in a two-way monitor at the moment.

WOBBLE BOARD

The rear panel wobbles a little because the amplifier and associated electronics are spring-mounted to avoid LF vibration at high volume. The rear panel gives you an XLR input, level trim and bass roll-off. Inside the amp module you get electronics designed and assembled by PSI with many handmade components — PSI is especially proud of the hand-wound tweeter.

G POWER

Power comes from an analogue Class G amp delivering 120W for the woofer and 50W to the tweeter.

Proprietary crossovers separate the drivers at 2.4kHz. They’ve got some great specs including 119dB max SPL at 1m, -6db down points at 38Hz and 23kHz, ±2dB tolerance between 44Hz–20kHz, and Total Harmonic Distortion is quoted at <1.4% between 90Hz and 12kHz.

black grille is available if fingers or flying objects are a concern. The dimensions are 250x400x300mm and weight is a solid 13.4kg. Some similar speakers are quite deep but these are a good size and easy to place. They look like modern furniture mounted above a mixing console; the finish catches the customer’s eye and the PSI logo on one side of the cabinet literally screams boutique.

3-LETTER WORDS

The heart of these speakers’ quality is more than skin deep, with a few PSI-developed technologies responsible for their supreme sound. AOI (Adaptive Output Impedance system) continually adjusts the amp’s output impedance depending on the frequency content of the program. This optimisation of the damping rate allows for extremely fast acceleration and deceleration of the speaker cone without overshooting at the top or bottom. PSI claim this system lets the speaker get close to reproducing square waves. To the ear, the sound is fast and accurate. An interesting side-effect of this system is its ability to sense and correct potential cone distortions caused by other sounds in the same environment.

CPR (Compensated Phase Response) uses all-pass filters across several frequency bands to present a phase-coherent output over a wide frequency range. This explains the rock solid stereo image for monitoring and ensures maximum phase coherence in multi-speaker systems.

While we’re on three-letter acronyms there’s also ALG (Acoustic Load Guide) that sees the tweeter set back in a little tunnel before flaring out at the front of the wave guide. The claimed benefits are wider frequency range, extended bandwidth and higher volume… and I hear them all.

LOUD & FIRED UP

SHARPEST TOOL IN THE SHED

When I fired up the A21-Ms with a few familiar tracks, the stereo image made a big first impression; wide and definite with a sense of almost seeing into the soundstage. Slightly mid-forward with amazing transparency, you get that feeling you’re hearing the tracks for the first time. These speakers will have you auditioning tracks just to hear what they sound like, as well as revealing sounds you hadn’t noticed before in more well-known works in your catalogue.

They starkly demonstrate the differences between recordings and easily highlight compression used in both recording and mastering. The details of distortion and reverb are clearly audible. The frequency response is wide and flat with not an ounce of flab in the bottom end. They look distinctive but don’t really have a sound of their own; they just reproduce what goes in with extreme accuracy. These are genuine studio monitors; no exaggerated or boomy bass, no scooped mids and no tizzy highs… just the truth. Starkly presented but beautifully delivered.

And they kick big butt when you turn them up.

This is another side of these speakers; they run loud for this size box, they respond to the volume knob like a car with a high-revving engine, and they stay nice right up until just before the limit light (red line) comes on. They bite if it’s in the music and can sound brutal because of this accuracy rather than in spite of its absence. They can also sound big, like a good PA, if they’re up loud and you’re right in front of them. You can sit as close to them as you like but about one to two meters in front seems ideal. Their 90 x 90° dispersion presents a wide listening area. In a good room they would have the power and throw to work well at medium distances.

In use, you forget about how good they sound and start to appreciate them as a tool of trade. I loved them for tracking; their crisp transients and transparency get you right inside the sounds. The power and volume give you the dynamics and detail you need for tracking drums and other loud sources. The tweeter is a cracker at all volumes.

And they’re perfect for giving the band a bit of a thrill on playback too.

Mixes I was already working on had good bits and faults equally well exposed. Early on, I was wrestling to control a harsh-sounding guitar band, but after a good tweak through the A21-Ms it all came together and I was very happy to hear how it sounded in the outside world. I proceeded with confidence; if you like the sound on these it will sound good anywhere. Once confident and familiar with them, both mixing and mastering benefited because I could hear everything I touched — these speakers reveal things to you. The only trouble I had with them was running them too loud, such a pleasure it was hard to stop, but with power comes responsibility, I suppose.

Quality touches abound. I like the silent off/on switching and it’s reassuring to know each speaker has been tested in PSI’s anechoic chamber and supplied with its own frequency plot. PSI aim to make the best monitors possible and to my ears these are as good as you can currently get. It’s not possible to be familiar with every speaker available, and personal choice comes into it sometimes, but these are right up there. There’s a price to pay for this sort of performance but it only takes a few seconds to hear what it buys. Used on a daily basis they are an ongoing pleasure.

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REVIEW

AwTAC 500 Series Compressor Last Issue we gave the AwTAC Channel Amplifier a big thumbs up.

This issue we examine its 500 series sibling, the unsurprisingly named Channel Compressor.

NEED TO KNOW

Review: Greg Walker

PRICE $1279 CONTACT Federal Audio: 0475 063 309 or sales@federalaudio.com.au

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PROS Great build quality & feature set Does transformer saturation & toughening-up of sounds very well Blend control and HPF give extra flexibility

CONS Three attack time settings only Fairly pricey thanks to the boutique build quality

SUMMARY A capable compressor that likes to be driven and excels in more hardhitting roles where the AwTAC’s love of transformer saturation comes into play. A well-made rock ’n’ roll device.

Compared to the comprehensive layout and feature bonanza of the double-width Channel Amplifier, AwTAC’s compressor is, at first glance, a much more conventional 500 series beast. The chunky faceplate sports industrial-style distressed-metal grey and silver chic and the build quality is exemplary — all through-hole, discrete components (apart from the metering) and knobs and switches that are smooth and responsive. The company’s NYC hand assembly ethos and passion for transformers and transistors is once again fully evident. The Channel Compressor is a FET-based design and is transformer balanced on the input and output stages. The side chain control element is placed after the input transformer so the input can be driven hard pre-compression, while multiple stages in both the input and side-chain circuits contain carefully tuned discrete transistor amplifiers (these also play a big part in the unit’s sound). Five rotary controls from top to bottom take care of input drive, gain reduction, release time, processed/dry blend and output gain. Alongside these are four small silver toggle switches that control a gentle high pass filter (6dB/octave below 375Hz) on the compression circuit, attack time selection (fast 1.5ms, medium 15ms, or slow 40ms), an auto release option and full hard-wire bypass.

Rounding out the front panel features is an eight LED gain reduction meter with a scale from -1/2dB to -15dB. DRIVE TIME

When I first fired up the AwTAC Channel Compressor I found a ready use for it in pumping up some fairly clean drum sounds I had recorded a few weeks earlier. I ran snare and kick sounds through one unit, sometimes gated and sometimes not, and shaped the transients on both these sources while adding weight and substance to the sounds. This approach worked a treat and I soon found the AwTAC to be a great comp for electric guitars and general drum bus duties as well as bass guitars and more aggressive vocals. There’s a lot of play in the gain structures with this box so you need to experiment. The clean gain reduction is very useable though quite subtle at lower levels, but when you introduce more drive on the input things get pleasingly gritty and thick. Above unity, the drive pot introduces distortion into the compressor circuit and is very effective at delivering anything from a subtle saturated effect to downright hairy palms. The HPF really plays a big role in how this compressor behaves tonally: engaging the filter creates a bigger-than-life bottom end image; while leaving it off allows the comp to really squish down the lower registers for a more pronounced midrange focus. The three position attack switch is a little limiting but works OK for most sources, while the auto release setting yields a program-dependant response that works very well on most sources. The Channel Compressor holds its tonal balance well under extreme duress and can also lend a nice ‘transistory’ thickness to program material on the stereo bus. The Blend pot is interesting and perhaps slightly misleadingly named. In fact this pot is a feed of the direct, uncompressed signal which can be blended with the compressed signal coming through the Output pot. Both these signals are routed to an internal 2x1 mono summing bus which combines them for your final output, so there’s lots of control there for parallel processing when you need it.

UNDER A-TAC

Overall, I thought the AwTAC Channel Compressor did its best work in more aggressive roles with the drive control pushing the input circuit hard.

For subtle and/or transparent dynamic control there are other 500 series devices that perhaps have a wider sweet-spot before saturation kicks in, but the AwTAC also performs well at more modest settings. This compressor definitely does a ‘thing’ and does it extremely well, particularly on rock and other harder styles of music where impact and a little or a lot of ‘hair’ on the source can make all the difference. With the bonus of true parallel processing and the ability to effectively dial out the compression and use it purely as a drive box, the Channel Compressor is a well thought out and tasty 500 series dynamic controller. If you like some grit in your sounds the AwTAC Channel Compressor should definitely be on your shortlist.

www.ADKMIC.com/thor

www.paservices.com

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REVIEW

ADK 3 Zigma Z-MOD Custom Shop Valve Microphones ADK wears its colour on its sleeve with the Z-Mod series of valve condensers, but it’s what’s inside that counts.

Review: Mark Davie

NEED TO KNOW

Everyone needs a bit more colour in their lives. And ADK is trying to scratch that particular itch. Its custom shop 3 Zigma Z-Mod series of microphones takes its standard Chinese-made multi-pattern valve condenser and pimps it up with some tasty tonal variations and metallic powder-coat colours.

The starting point is ADK’s TT multi-pattern tube condenser, which has a vaguely AKG C12-ish/Telefunken ELAM251-ish body. It has a dual-layer grille head basket, and a straight body that easily screws apart so you can take a look inside or swap out tubes.

The Z-Mods upgrade the cosmetics with

PRICE RRP$3799 (call for pricing) CONTACT Professional Audio Services: (02) 6059 1652 or sales@paservices.com

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chrome head baskets, and powder-coated bodies in a range of appealing colours. And in a show of quality control, all the Z-Mod mics screwed apart and fit together more confidently than the standard TT. The lineup comprises five microphones that get their names from various classic microphones — there’s the Z-12, Z-47, Z-49, Z-67 and Z-251. No prizes for guessing where those numbers came from.

They all use the same basic body, PCB layout, and custom power supply. Overall, the quality of components has gone up a few notches over the standard TT — bigger capacitors and

PROS Plenty of character across the range Well-made with quality components Multiple tubes to choose from Noise free & no nasty resonances

CONS Have to really crank suspension mount locking mechanism Serial number sticker a bit cheap

resistors, better transformers and tubes. Between the Z-Mod mics, there are a few resistor value changes, and different transformers — you get a Lundahl LL1530 in the Z-49 and Z-67, a Jensen in the Z-251 and Sowters in the Z-12 and Z-47.

All the mics come with a selection of JJ and Mullard tubes to play around with in the 12AX7/ECC83 family, a step up in gain from the Chinese-made 12AT7 in the TT. The tube type is the same across all mics, so you can swap out tube flavours even if you have different models — even though the Neumann U67 uses an EF86 pentode wired in as a triode, and the ELAM251 used a 6072/12AY7, which has much lower gain

SUMMARY There’s a touch of vintage soul in every ADK 3 Zigma Z-Mod mic, with more modern specs to boot. The looks aren’t just skin deep either, with plenty of attention to detail and quality components inside. If you’re after a classic sound, but want security in your purchase, for once, a mic in a Chinese body is a great way to go.

than a 12AX7. It seems circuit designer, JP Gerrard, is making his work easier by keeping the PCBs uniform, but it also means these aren’t dedicated copies — a good or bad thing depending on how influenced your ears are.

TIME CAPSULES

The main difference in each mic is the capsule, designed by a mysterious Australian aerospace engineer, though ADK has managed to keep their identity buried. He or she is like the Stig of audio.

Or perhaps ADK thought no one would bother looking for the engineer Down Under.

While the capsules might not always physically resemble their vintage counterparts, sometimes that’s for good reason. For instance, rather than cloning the U67 capsule — requiring ADK to implement a new PCB layout with an equalisation circuit like the original mic — the capsule is designed to mimic not only the response of the capsule but the frequency response changes of the U67 circuitry. It’s an all-in-one 67.

The normal TT is no slouch of a microphone.

It’s Chinese-made, and comes stock with a Chinese made 12AT7, as opposed to the higher gain 12AX7.

It comes with essentially the same accessories, bar the souped up custom power supply and higher

quality cables. It’s a nice-sounding mic, with a bit of a boost around 2.5kHz that sounds a bit throaty and pushes sources forward. But the limitations of the capsule and electronics have been revealed by the specifications quoted on the Z-Mod series, showing the upgraded components are at least having an effect on performance. The ADK Area 51 TT has an EIN spec of 20dBA, the Z-MOD mics are 16dBA and below, and the Z-Mod mics handle a few more dB of SPL. They’re still relatively sensitive though, and while they are useful for recording most sources, you’ll want to have pads on hand for anything loud.

It adds more full round bottom end on all sources without ever losing clarity. It also balanced out the top end boost, giving it more of a foundation.

The rest is just a matter of flavour and where you want a source to cut through. If you just want more top end — or really, less bottom — go for the Z-67. I found this was the best fit on acoustic dreadnaught-shaped guitars, the gentle rolloff was perfect for the low end, where the Z-251 was too full. A more neutral top end? Unpack the Z-47. If you want more harmonic bite, have a look at the 49, especially on percussion, I liked what it did to the high end.

SOUND CHARACTER

SOUND BODY, SOUND MIND

I was able to spend some quality time with four of the Z-Mod series, only missing out on a date with the Z-12. With no frequency plots on hand, it was a fun game to really stretch the ears and listen for the subtle differences. Depending on the source, these could be quite subtle, but after a lot of listening, I felt I was able to grasp the character of each mic and what it would do for me.

After doing all that, I was pointed to the capsule plots online by Adam Boon from Professional Audio Services. And it was intriguing to see if what I was hearing was in the charts.

The main noticeable difference between all the Z-Mod microphones and the TT was in the high end. All of them had more air up there.

The Z-251 had generally better reproduction down low that rounded out most sounds and gave them a sense of fullness. I felt that it had a little more pronounced low mid around 180200Hz on some sources, but it was really just the low end extension I was hearing. Interestingly, the Z-251 also has the most boost in the high frequencies, whereas most of the other capsules start rolling off above 15kHz. But to be honest, I wasn’t hearing much of a difference on most of the sources I was recording.

My sensation when comparing the Z-67, was that it had more top end across the board, with a slight general lift centred around 4kHz. It was nice to see this confirmed on the frequency plot, my ears doth not deceive me. It was interesting to see how wide a boost it had though, from about 2-6.5kHz. And the overall feeling of hearing more top end probably also had a bit to do with the gentle roll off starting at about 200Hz as well.

The Z-47 was probably the most neutral of the lot. I kept coming back to this one on male vocals, it just seemed to be the one that captured the sweetest upper mid range.

The Z-49 sounded like it had a bit of a bump around 1kHz, which gives a nice round harmonic, washy aggression to things like cymbals without being overstated. Really, it just made the mic sound nice and full in the areas where you want it to push.

If you had to twist my arm, I would say the Z-251 would probably be my favourite. In the way we often describe tube gear as harmonically rich and warm, the Z-251 most lives up to that ideal.

For a standard Chinese mic body, it seems to be pretty sound. There wasn’t any really noticeable basket resonances, given the variety of high frequency responses I was hearing from the different capsules and electronics.

The suspension mount works pretty well. It’s a simple design that feels robust, although the locking angle adjuster could do with a refresh, as the mic’s weight pushes the boundaries of its grip and you have to really crank it.

The packaging is also neat, the whole kit packs away nicely in a metal case, and the microphones have an extra layer of protection courtesy of a velvet-lined wooden box. The power supplies are nice, with nine polar pattern selections from omni, through cardioid to figure-eight. The rejection in figure-eight was very good, and the back side often seemed to have a slightly duller tonality to it, which might come in handy. The cardioid setting is not super wide, but you’ve got plenty of scope to spread it wider or thin it down with the polar pattern selection.

The only thing letting down the level of finish is the stick-on label for the serial and model numbers.

It cheapens the whole look, and if it peels off, you’ll have to remember which flavour it is by the powdercoat colour.

NOW WHICH COLOUR?

These mics are really worth checking out if you’re in the market for a multi-pattern tube condenser. If you’re worried about the quality of Chinese-made condensers and were hoping to steer clear, opening one of these up should waylay any concerns — they’re full of quality components, and are well put together. Plus, the capsules and tubes are fitted and tested in the good ol’ US of A. They also come with a five-year warranty.

The differences between the mics is sometimes subtle, when you first put them up they all sound really great on a lot of sources, but with a bit of listening you can start to appreciate each for the character they bring. And while they’re not component for component remakes, these will stand up alongside any other tube mic you’ve got, and spec-wise, they’ll likely clean up any vintage comparisons.

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REVIEW

ZOOM TAC-2 DESKTOP AUDIO INTERFACE Zoom puts a Thunderbolt into the two-channel interface market.

Review: Mark Davie

I’m just going to make this easier on all of us: on the surface of it, Zoom has made what looks like a little Apollo Twin. Got that image in your head? Two-channel desktop interface with a single big knob, folded metal top plate and a Thunderbolt connection. Phew, glad we got that out of the way. Okay, let’s begin.

BOLT OUT OF THE GATE

NEED TO KNOW

The bolt side of the Thunderbolt is where Zoom is getting most of its bang for buck — bolt of speed, and a bolt of power.

The big sell of all the Apollo interfaces, including the Twin, is the ability to track

PRICE $685 (expect to pay $550) CONTACT Dynamic Music: (02) 9939 1299 or sales@dynamicmusic.com.au

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through UAD plug-ins using onboard DSP — latency through your DAW doesn’t really matter with Apollo. But for the Zoom, and any other desktop interface, latency is a big deal.

Why? Because most users will be using it to track everything from a DI with guitar amp simulators to vocals with onboard effects — and lag is a killer for tracking.

On the face of it, Thunderbolt’s impact on speed would seem negligible. Most twochannel interfaces run over USB 2.0, which is more than enough bandwidth for stereo I/O, even 24-bit/192k. But Thunderbolt’s 10Gb/s transfer rate seems to be doing its job here.

PROS Thunderbolt bus-powered Low latency, great for tracking Handy auto-gain feature actually useful Preamp & conversion good for this level

CONS No Thunderbolt thru port

Latency on a two-channel device such as this is down to the stability of the driver — especially when it’s under processing load. The TAC-2’s roundtrip latency (using the monitor outputs) was a low 132 samples (2.75ms at 48k), plus double whatever buffer you’re using. In practise, when recording direct into one of the inputs, latency was negligible for buffer settings up to 64 samples with a decent chain of plug-ins inserted — little enough latency to get intimate with compressed, effected vocals, and speedy enough to rip out virtuoso solos through an amp simulator without feeling behind.

The other side of it is bus power, Thunderbolt

SUMMARY Zoom might have packaged this up to look like a mini-Apollo, but it’s the low latency and Thunderbolt bus power that makes the TAC-2 a quality contender for your desktop needs.

delivers 18V DC over the cable, compared to USB 2.0’s roughly 4.75V. More power means more choice, and in this case Zoom has opted for a higher spec preamp chip than you’d expect. I’m saying that in all deference to Zoom, as I’ve used its guitar pedals as a wet-behind the ears teen and I’m intimately acquainted with its portable recorders.

Over the years Zoom has always packed in plenty of features for the price. Functionality has also been a strong suit for Zoom, and lately the development of the portable recorder range has shown a commitment to developing the sound and build quality side of the business too. It shows in the TAC-2. While the plastic, push-button knob isn’t the most confidence-inspiring piece of hardware, the metal faceplate, metering and balanced connectivity are all solidly fitted.

So what was that preamp? Well, it’s a Burr-Brown PGA2505, it’s slightly noisier than the PGA2500 chip you’ll find in Antelope, Prism, RME and other top manufacturer’s gear, but it has many of the same digitally-controlled feature set and sounds quite good. The Thunderbolt bus provides enough to power both channels and the headphone amp, which a USB bus would struggle to do. I do have better preamps around, but I’d have no problems tracking through the TAC-2 — it’s clean, handles low end well, and only gets noisy when pushed into silly territory.

ROUND TRIP

The TAC-2 is obviously Mac-only at this point, with Thunderbolt not really making any headway in the PC market. It’s also limited to stereo I/O, so there’s no separate headphone and monitor outputs, which you do get on the Apollo. I don’t see this as a big deal considering the style of tracking being done with this kind of interface. There’s also no ADAT or similar expansion port. Again, this makes sense for this device. While the headphone amp doesn’t go super loud like the Apollo, it’s clean and I found it suitable.

In the box, you get a metre-long Thunderbolt cable, something you don’t get with the Apollo, and a $20 saving. There’s still no through port for

Thunderbolt though. This is not a criticism of Zoom alone, it’s true of most Thunderbolt devices.

It’s particularly annoying if your Macbook only has one port and you want to run an extra screen. But the big upside is the Thunderbolt bus-power that eliminates the need for a separate power supply.

CONTROL PANEL

The software control panel is very well thought out and simple to use. There are even arrows tracing the signal flow so you just have to follow your nose. The direct path links both preamp sections into the computer, the preamps also feed a mixer section where you can create a balance between the inputs and your computer’s playback for a monitor mix if you want zero latency monitoring, or have to raise your buffer settings. From there, you can set the level of your monitors or headphones and incorporate global reverb using a send control for each input and the computer playback. There are eight reverbs to choose from with two from each of Room, Hall, Plate and Echo.

The oddest function on the control panel is Loopback — a throwback to a time before desktop interfaces even existed. The gist is, instead of recording both channels direct, you tap the monitor mix instead, which can be any combination of both tracks, plus the computer playback and effects.

The difference is, it all goes back into one mono or stereo track. The only real use I could see for this is laying down an initial bed track where you wanted to blend two mics you liked the sound of (say a ribbon and dynamic on a guitar amp), plus the reverb if you were particularly smitten.

I say ‘initial’ because the monitor mix also feeds your headphones, so any amount of the computer playback you dial into your monitor mix will also be recorded on top of your new tracks in Loopback mode. Honestly, I’ve been over it a few times, and for the life of me I can’t really work out what you’d want this feature for unless you were particularly sold on the Zoom’s reverb quality — send me suggestions on a postcard.

Back to the parts that make sense. The metering is really solid, with a wide bar graph meter for

everything, including computer playback and separate meters for headphones and monitoring.

There are three preset memories, which are handy, dials and buttons for everything else — low cut filter, phase flip, and phantom per channel. There isn’t a pad option but the gain pot does have a unity gain setting.

GAINING BIG

The TAC-2’s gain range stretches from +9dB to +60dB, which is wide enough for most sources.

However, one of the downsides of using the slightly cheaper Burr-Brown chip is that gain is set in 3dB steps.

Always sticklers for interesting functionality, one of the neatest things about Zoom’s device is the handy Auto-gain setting. By pressing the Auto button in the control panel and feeding it your signal, both channels simultaneously try to find the optimum gain setting for the material. You can also set the highest input to target 0dB, -6dB or -12dB — just bash, strum, or sing away and let the Zoom dial it in. I’ve slowly grown to appreciate auto gainstaging — having come across it more frequently on digital consoles — and the Zoom version works particularly well.

The hardware knob also lets you flick through setting levels for channel gain, monitor and headphone outputs with the knobs push-button selection. One of the stops allows you to adjust the two channel gains together if you have a stereo pair of something, or if you just want to raise or lower the overall system gain. It’s a nice touch, and wellimplemented.

The Zoom TAC-2 is a really capable device. It might not carry the ‘UAD on-board’ designation, but with its super-low latency, you can track through your DAW’s plug-ins with no issues. If you’re after a good-sounding, two-channel interface for your Macbook that’s fast and without breakout cables, then this little unit could be the answer.

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REVIEW

SHURE QLX-D WIRELESS Welcome to the latest mid-level addition to Shure’s digital wireless family. Expect some analogue cousins to be shown the door.

NEED TO KNOW

Review: Christopher Holder

PRICE QLXD24SM58 (w/SM58 Handheld): $1823 RRP QLXD1483 (w/Lavalier): $1939 RRP CONTACT Jands: (02) 9582 0909 or sales@jands.com.au

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PROS Many pro features at a semi-pro price Easy to set up Shure’s rechargeable batteries CONS Some won’t like the handheld’s on/off switch

SUMMARY The Digital Dividend doomsday clock is ticking and QLX-D is a good fit for those who intend on ‘buying up’ (replacing a motley crew of single wireless channels) or ‘buying down’ (where QLX-D’s features are perfectly sufficient and it’s a case of ‘why pay more?).

SHURE’S NEW DIGITAL HIERARCHY Frequency Band PGX-D: 902-928 MHz* GLX-D: 2.4 GHz QLX-D: 470-814 MHz* ULX-D: 470-814 MHz* *Country Dependent Max. Simultaneous Systems PGX-D: 5 GLX-D: 4 QLX-D: >60 ULX-D: >100 using high density mode

Latency PGX-D: 3.5ms GLX-D: 4 ms minimum; 7.3ms maximum QLX-D: 2.9ms ULX-D: 2.9ms Sampling Rate PGX-D: 48k GLX-D: 44.1k QLX-D: 48k ULX-D: 48k All use 24-bit sampling resolution.

Max. Operating Range PGX-D: Up to 65m GLX-D: Up to 32m indoors QLX-D: Up to ~100m ULX-D: Up to ~100m using standard density mode

Shure’s wireless range is going through a bit of a ‘restack’ of its own. 24-bit digital is moving in, pushing legacy analogue systems out of the way — from the bottom of the range to somewhere near the top (with ULX-D doing great business).

QLX-D sits in the ‘upper mids’ of the Shure wireless range. The big drawcards are its wide tuning bandwidth (64MHz), the fact you can network multiple receivers via ethernet (for one-button frequency finding/selection), and its compatibility with Shure’s SB900 AA recharge battery pack.

If like me you’ve not totally covered yourself for the onset of the spectrum squeeze next year, then pay attention. If you’re a mid-sized performing arts centre, a biggish church, university, PA rental company or take care of sound for a government department or conferencing centre, then QLX-D delivers enough features for you to avoid spending the really big bucks.

PLAYING WITH THE PROS

As someone who’s not invested in wireless for a couple of years, the QLX-D has been somewhat of a revelation. For around $1600 (street price) a channel you get a system that’s suitable for just about every application, short of missioncritical musical theatre, festival sites, or royal visits. You can put dozens of receivers in your rack before it complains; with Shure’s Wireless Workbench software you can have pro-grade wireless admin and oversight; and with the ShurePlus Channels Mobile App you get remote monitoring (RF and battery status for starters) from your iPad or iPhone.

Software like Workbench has traditionally been the preserve of the RF guy, ie. on gigs that warrant a specialist RF technician. So to have this kind of functionality in a semi-pro system is a luxury and, again, means mid-level users who might have a

‘Christmas Special’ or a one-off large conference can respond with QLX-D without the need to hire in heavier artillery.

A STEEL

This system is a pleasure to operate and work with. The hardware itself, with the steel cases, feels sturdy and less ‘consumer’ than, say, my trusty SLX4 units. Saying that, in my experience even Shure’s plastic ‘bits’ that feel flimsy never fail anyway — but the steel construction is reassuring.

The handheld mic is nicely weighted. It’ll slot straight into the optional recharge station, or you can feed AAs into it. Somewhat controversially, the handheld has a mechanical on/off switch, rather than a more tamperproof press-button. In practice, I’ve never had an accidental power-down — which makes me happy; while feedback from talent has been universally positive — they love the big on/ off switch. For that extra level of ‘high-wire act’ difficulty it does allow performers to manually switch the device on and off depending on whether they’re on-mic — as I recall Jamiroquai does in his live show — to assist FOH with stage spill pouring down the microphone. I’ll leave it up to souls braver than I to recommend this particular technique.

The half-rack receiver is, again, solid and workmanlike in appearance. The LCD is bright and easily viewed from any angle. The battery life meter is an utter godsend. The number of ‘probably okay’ AAs I’ve jettisoned over the years is shameful, and having that ‘hours:minutes’ visual feedback I think lowers the levels of my anxiety from Defcon 1 down to a positively sedate Defcon 5 on longer shows, or when I’ve forgotten to run a multimeter across my half-used batteries.

Finding the best frequency and syncing the receiver (via IR) to the transmitter is child’s play.

Even better, as mentioned, you can network the receivers: plumb in an ethernet switch and QLX-D AT 69

DIGITAL WIRELESS Digital wireless means different things to different manufacturers. For many they’re referring to digital controlled analogue (meaning the signal still travels in the analogue domain via RF). Shure QLX-D and ULX-D both use true digital transmission.

The trouble for Shure (and others) is that a channel of 24-bit/48k audio occupies more bandwidth than an equivalent analogue signal.

‘Well, give Fraunhofer a buzz and apply some kind of data compression,’ I hear you say. Nothing doing, as that sort of jiggery-pokery introduces unacceptable latency.

The answer? Shure has patented a quadrature modulator system that provides the advantages of increased signal-to-noise (120dB), a more efficient use of the bandwidth, and a greater tolerance of other active systems in its near vicinity.

If you want some bed-time reading tap ‘Shure digital modulation’ into Google Patents for more.

will automatically take care of the channel scan and frequency configuration for all the receivers in your rack — no need to get a PC involved.

The review kit shipped with an SM58 head and a WL93 lavalier system. The 58 worked as expected.

You can select an SM86, Beta 58A, Beta 87A, Beta 87C or KSM9 capsules. I would have been interested in testing the MX153 earworn headset mic, so it’s worth noting there is that option, along with more primo lavalier choices. There’s an instrument/guitar beltpack transmitter as well.

QLX-D converts audio to 24-bit/48k before transmission [see the Digital Wireless box item for more detail]. There’s 120dB of dynamic range, a number that would be like science fiction only a few short years ago. You want reduced dynamic range? Apply compression at the desk — no need to gain up or down the transmitter to find a sweet spot. I had vocalists of all descriptions — from hoarse whisperers to hold-onto-your-hat belters — using the mic and it performed flawlessly. Did it suit every vocalist’s tone? Of course not. But the clarity and performance was evident.

stronger 20W transmitter power. There’s also Shure’s ULX-Pro analogue system, with QLX-D almost certainly taking much of its market share — only the die-hard ‘analogue or nothing’ wireless stalwarts will insist on ULX-Pro. Meanwhile, if you’ve got more than a handful of wireless channels and you find yourself spending, say, 30 minutes before every gig configuring the system and checking/replacing batteries, you deserve to upgrade to a system like QLX-D. Anything cheaper is false economy.

Anything more expensive, could well be unnecessary for all but those ‘failure is not an option’ applications I mentioned earlier.

I’ll certainly be giving QLX-D serious consideration. The Digital Dividend countdown clock is ticking and I like the idea of investing in wireless that can be complemented as budget permits, becoming a true system rather than a rag tag bunch of individual channels.

CASE FOR QLX-D

As I alluded to earlier, QLX-D shares much with the pricier ULX-D — same digital transmission, and anecdotally, the same sound quality — but ULX-D brings pro features such as Dante, dual/ quad-channel receivers, and the option of a

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REGULARS

LAST WORD with

Billy Woodman, Managing Director of ATC Loudspeakers Photo: Hugh Hamilton

2014 marks the 40th anniversary of ATC speakers. Head designer and owner, Billy Woodman, was first interviewed in Issue 1 of AudioTechnology.

In that time, ATC has come full circle, from public address, to studio monitors, into hi-fi, and recently has returned to its public address roots.

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I suppose there were two areas of inspiration for me.

One, the original JBL products, because they used to make a lot of short coil/long gap design-based drivers when they first began in the ’50s and ’60s. People that influenced me greatly were Quad’s Peter Walker, Laurie Fincham and all that group of engineers that came through Goodmans.

In 1970, I went from Revolver in Melbourne to England, because I’d written to Goodmans loudspeakers and got a job with them. I played piano on a boat to get myself over there.

I learnt more about loudspeakers in my four years at Goodmans than during any other time. It was the biggest loudspeaker manufacturer in Europe — it had a huge laboratory and people there with a wealth of knowledge.

While I was at Goodmans I did a Master of Science in Acoustics at London University. In 1974, I started ATC with one 12-inch public address loudspeaker. I’ve been doing ATC ever since. It turned 40 in May. It’s still small — 26 people — but we’re one of the biggest specialised loudspeaker manufacturers in the world. It’s what’s kept ATC alive; we make a series of highly specialised components that you can’t buy over the counter.

We’ve been able to design for performance first and then work out whether it’s viable to make. Whereas if you go to large component manufacturers for your supply, you’re stuck with what they’re making for big manufacturers — that’s the reality. Just about everywhere else has almost given up making transducers; they’ve become system manufacturers.

ATC is the only truly, vertically-integrated loudspeaker company that’s left in Britain. We can still do everything, whereas KEF and Celestion have all gone offshore.

When it’s all said and done, the maths behind loudspeaker design has been firmly established for a very long time. What’s changed is the ability to make digital measurements. Before, you had to use a lot of intuition to interpret the results. Now you can see it just through taking measurements.

All we’ve ever done is try to better engineer what’s already known. The only thing that was relatively new was the superlinear magnet. But it wasn’t patentable because it’d been used in the communications industry.

I stumbled across it by accident. I’d been looking at laminating poles to break up eddy currents and noticed a book on communication ceramics. Reading it I found someone had developed a material which was powdered iron with an oxide coating compressed into a ring — it had good magnetic permeability but high electrical resistance. Using it knocked the odd harmonic distortion down by 12-15dB.

It’s only really useful if you’re using a short coil in a long gap. Most people go a long coil in a short gap because they’re cheaper to make, but there’s no question: if you’re looking for linearity, a short coil in a long gap is by far the best solution to the problem.

In the ’90s, the big PA companies were no longer driven by performance, but cost. Cheap components were readily available from various sources in the Far East. So it meant manufacturing high-spec drive units for the PA industry — how ATC started — was going to lead us nowhere, so we continued diversifying into home theatre.

Being in the business a long time, you make sure you never let the grass grow around your feet for too long. Since 2004, we’ve been doing large auditoria.

Disney Hall opened in 2004, with acoustics by a chap called Yasuhisa Toyota of Nagata Acoustics. The sound reinforcement comprised large line arrays, which were not really appropriate for a high performance auditorium. It was brought to a head when pianist Keith Jarrett said in an L.A. Times article it was the worst auditorium he’d ever performed in and wouldn’t come back!

It’s not a good idea to use line arrays when listening to music performed onstage in an auditorium because of the psychological disconnect between your visual experience and the source of the sound. The distortion and directionality alone will drag your attention to it, because its pattern is entirely different to the pattern you’re hearing from the stage.

Although an auditorium is a large, reverberant space, the RT is usually well controlled from about 50Hz to 2-3kHz. As a consequence, you can treat it like you would a recording studio control room.

A friend of mine did all the recording at Disney Hall and dragged me in. We happened to have SCM150 monitors on hand because Pink Floyd had been launching the SACD version of Dark Side of the Moon all over The States using five SCM150s and two subs.

We did a trial run where we simply mounted 150s on top of one another either side of the stage. They ran it for four months and the criticism disappeared, so they gave us a contract to build a system for the auditorium.

Frank Gehry and Nagata Acoustics took us on the next project, which was a new auditorium for The New World Symphony in Miami. We also got the Bing Concert Hall at Stanford University, and are just about to finish the new auditorium for the Shanghai Symphony Orchestra. All as a consequence of being at the right place at the right time.