AudioTechnology App Issue 25

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Introducing the new NTR active ribbon microphone from RĂ˜DE The finest ribbon microphone ever made.

Hear it for yourself at rode.com/ntr

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Editor Mark Davie mark@audiotechnology.com.au Publisher Philip Spencer philip@alchemedia.com.au Editorial Director Christopher Holder chris@audiotechnology.com.au Editorial Assistant Preshan John preshan@alchemedia.com.au Art Director Dominic Carey dominic@alchemedia.com.au

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

Graphic Designer Daniel Howard daniel@alchemedia.com.au Advertising Philip Spencer philip@alchemedia.com.au Accounts Jaedd Asthana jaedd@alchemedia.com.au Subscriptions Miriam Mulcahy mim@alchemedia.com.au

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

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

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All material in this magazine is copyright Š 2015 Alchemedia Publishing Pty Ltd. Apart from any fair dealing permitted under the Copyright Act, no part may be reproduced by any process with out written permission. The publishers believe all information supplied in this magazine to be correct at the time of publication. They are not in a position to make a guarantee to this effect and accept no liability in the event of any information proving inaccurate. After investigation and to the best of our knowledge and belief, prices, addresses and phone numbers were up to date at the time of publication. It is not possible for the publishers to ensure that advertisements appearing in this publication comply with the Trade Practices Act, 1974. The responsibility is on the person, company or advertising agency submitting or directing the advertisement for publication. The publishers cannot be held responsible for any errors or omissions, although every endeavour has been made to ensure complete accuracy. 10/11/2015.

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COVER STORY

16

Tame Impala’s Current Nostalgia

ISSUE 25 CONTENTS

28

How Do You Measure Up? Part II

Studio Focus: No Wave Studio

11

MOTU Joins the AVB Network

Mackie ProFX V2 Mixers AT 6

42

Jamie xx: Hits the Spot with In Colour

36

Rode NTG4+ Shotgun

22

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SSL Sigma Analogue Summing Device 44 28

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GENERAL NEWS

AKAI MPC: NOW WITH TOUCHSCREEN The Akai MPC — a familiar face in the music production world — has had a face-lift. The Akai MPC Touch now features a seven-inch colour, multitouch display, along with MPC’s improved backlit pads. Multi-touch capability means you can reach out to grab and pinch waveforms, draw MIDI events, adjust envelopes, chop samples, add effects and precisely set controls using your fingertips. The MPC Touch also features a step sequencer, phrase looper, pad mixer for setting levels and stereo panning, sample edit control for processing your

samples, data encoder knob for push-and-twist control of display parameters, and four new performance-ready touch-sensitive control. MPC software and over 20,000 sounds are bundled with the controller. The controller and software is compatible with Max OS X 10.8.5 or later, and Windows 7 or later. Expected to be available in Australia from late November.

Electric Factory: (03) 9474 1000 or www.elfa.com.au

BAREFOOT MICROMAIN 26 Barefoot Sound has added to its line of high-end monitors with the MicroMain 26. The MicroMain 26 is a four-way active system with six drive units housed in sealed enclosures, with the 2.5-inch aluminium cone midrange driver nestled in a 3D-printed waveguide enclosure. The frequency range is quoted as spanning 30Hz-45kHz, and the Multi Emphasis Monitor Emulation (MEME) Technology enables users to switch the MicroMain 26 from the revealing ‘Flat’ response to the

warmer ‘Hi-Fi’ setting, NS10-emulating ‘Old School’ setting, and ‘Cube’, which models the classic mix cube speakers. The crossover design took Thomas Barefoot more than four years to develop — with the crossover points at 100Hz, 800Hz and 4kHz — and the amplifier stage was designed with Bruno Putzeys of Hypex to deliver transparency.

The Barefoot MicroMain 26 will be shipping in December 2015.

AUDIO-TECHNICA BP40 The BP40 is Audio-Technica’s first large-diaphragm dynamic broadcast microphone — specifically made for applications such as radio, overdubs, postproduction, podcasting and more. Its hyper-cardioid dynamic capsule is designed to provide good off-axis rejection and isolation, while maintaining a rich, clear and natural ‘condenser-like’ delivery. Audio-Technica has also fine-tuned the capsule’s placement to help maintain a solid vocal presence even at a distance, while its multistage windscreen provides internal pop filtering. A switchable 100Hz high-pass filter is also built-in to roll out any rumble. The already

large diaphragm, has a patented floating-edge construction designed to maximise surface area and optimise the diaphragm’s overall performance, while a humbucking voice coil prevents electromagnetic interference from video monitors, AC power, etc.

With a rugged all-metal construction, the BP40 is well-prepared for use in professional broadcast environments and comes standard with the AT8483 mounting clamp. The optional AT8484 shockmount is also available should additional isolation be required.

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

SLATE RAVEN MTi2 The Slate Raven MTi2 features all the functionality of its Raven predecessor but with a smaller price tag. The 27-inch, 1920x1080 production console gives you complete multi-touch tactile control of your DAW and now includes the new Raven 3.0 Control Software. The new software adds a user-customisable multi-function Toolbar and Batch Command System to save time and enhance the workflow of the audio engineer. Slate Remote is a free app for iOS users available to anyone

with the Raven MTi2. The app synchronises with your session via Wi-Fi to give you access to your batch commands. Most importantly, Raven 3.0 is compatible with every major DAW including Pro Tools 10-12, Logic Pro X, Ableton Live 9, Cubase 8, Nuendo 7, Studio One 3 and Digital Performer 8 and 9. It’s currently only compatible with Mac OS X but support for PC is expected soon.

Audio Chocolate: (03) 9813 5877 or www.audiochocolate.com.au

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AT SUBS PRIZE WINNERS

David Rogers from Reservoir in Victoria recently won our subscription prize, taking home a brilliant AEA Nuvo N22 phantom-powered ribbon mic. Greg Walker reviewed the N22 in Issue 107, describing it as “a thoroughly modern mic made by people who love vintage gear.”

David: “I’d like to thank AudioTechnology, AEA and Mixmasters for sending me the AEA Nuvo N22 ribbon mic. I've been making my own records for 20 years on cassette four-tracks, stand-alone recorders and now computers but it's only in the last year or so that I've branched out to make recordings for other people. I've been wanting to add a ribbon microphone to my collection but have always had other priorities to take care of in my studio. I can't wait to try the N22 on guitar amps and as a mono drum overhead. I've already used it to track some acoustic guitar and the slightly darker tone helped it sit beautifully in the mix.”

The

QUICK MIX

Based in Fremantle WA, Peter Oats started Kitchen Cooked Records more than a decade ago. He recently won a pair of killer Rode NTR ribbon mics. Being a longtime Rode user, was pretty keen to press the prizes into action.

Peter: “I have a pair of each Rode K2, NT5 and NT1 and love them so the NTRs will go straight to work! I love capturing great sounds and I realised early in the piece a great performance only comes from artists that feel comfortable.

I try my hardest, no matter who I'm recording, to get them in the zone and give me a piece of magic.

“I’ve been fortunate enough to work with people like Lucky Oceans, a twice Grammy Award-winning musician and David Milroy, artistic director for Yirra Yaakin Aboriginal Theatre Company.

“My current project is creating sound design for an e-book of amazing poetry and music written and performed by musician and private investigator Paul Thompson called Through the Eyes of a Spy. Hopefully my recordings have improved since the days of my tangerine e-Mac and Pro Tools Free!

with

Jamie Wellwarth Interview: Neil Gray

Who have you been touring with?

I’ve been mixing Jane’s Addiction, and I’ve also worked with Jimmie Vaughan, Doyle Bramhall II, Fitz and the Tantrums.

When did you start doing live sound?

About 1998 I started throwing punk rock shows in my friend’s basement and ended up owning a PA, so I wound up running the sound.

What’s your favourite console and why?

Currently it’s the Avid Profile — only because I use it all the time, and it sounds decent.

Favourite microphone?

I am a big fan of the Telefunken mics.

Most memorable gig or career highlight?

Mixing Jane’s Addiction on the Nothing Shocking tour was a real dream gig. I also mix at the Continental Club in South Austin and on any given night we will have some of the best musicians the world has to offer, it’s really amazing!

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How has your mixing setup changed in the last 15 years?

Well, I used to mix only analogue with a lot of cool outboard gear, now I pretty much use digital on the road because it’s easy. I keep my files on a USB stick, and all my plugins are cut up depending on what I’m mixing.

What game-changing pieces of gear have come out in the last 15 years?

Mixing on digital desks has really saved the FOH/monitor footprint, plus they are sounding better and better. I still use a Portico II Master Buss Processor most of the time for my left and right mix, it sounds amazing and really tightens up the mix with a wide stereo field.

Any tips/words of wisdom for someone starting out?

Don’t give up, if this is what you want to do you’ll have to put up with a lot of intern jobs wrapping cables, putting away mics and stands, or pushing speakers around.

Also, pay attention to engineers when they are talking about how they mix and try it out for yourself. Always be willing to learn and don’t think you know it all!

REMIX COMPETITION PRIZE WINNER BEN ALLEN With the top prize worth over $13k, the Remix Comp from AudioTechnology Issue 100 was one of the biggest we’ve ever held. Ben Allen (aka Broadwing) from Alice Springs, NT, claimed the blue ribbon, with the judges rating his mix of The Occupants’ Wonderland the hard-fought winner from a shortlist of deserving entries. The takings included a Pro Tools HD Native Thunderbolt system with an Omni interface, Avid Artist Control, Event Opal monitors and three Rode microphones — a K2 and a matched pair of NT5s.

It was a lot to get his head around, so we let Ben get on with it and bed in the new setup. Just over a year later, we asked Ben how he’s going with it all.

Ben: “It’s definitely legitimised my operation a bit more. That Omni interface is just streets ahead of what I was using, and the K2 valve mic matches my voice really well so it’s become my main vocal mic.”

Not just a step up in audio quality, the upgrades have also increased Ben’s confidence in his work: “I think I just feel more certain that things are going to translate and that what I’m doing is going to sound more like a professional product. I was exclusively monitoring with headphones before, and was never entirely sure that what I was doing was going to translate to other systems. It’s given me more confidence in the way I’m mixing.”

The first two songs Ben released with the new production system saw him take out the 2015 NT Song Of The Year award in two categories, which is a bloody good strike right. He attributes the success in part to the professional edge lent by the new gear. We’d give more credit to the gear,

but Ben’s no stranger to first place; he took out the 2012 APRA Tropscore film scoring award too.

He’s definitely a homegrown talent, making it even more of a pleasure to partner with Avid, Rode and Event to help elevate his productions. He released his EP Pinhole Camera last year, and a full-length follow up recording is in the works, giving the rig some good mileage.

Looking ahead, Ben hopes to wear as many hats as possible: “I want to do a bit of everything — more production work, more film scoring, and try

and write as many songs as I can as well.

“Cheers for running the comp, cheers for picking my entry and cheers to all the people who supplied prizes.”

Thanks again to Avid, Rode and Event for the awesome prizes, The Occupants and The A&R Department for supplying such a great track to work on, and our esteemed judging panel — Forrester Savell, Francois Tetaz, and Tom Larkin.

STUDIO FOCUS:

NO WAVE STUDIO 83 Centennial Ct, Byron Bay NSW nowavestudio.wordpress.com Producer/Engineer Julian Abbott has opened up No Wave Studio in Byron Bay around a SoundWorkshop Series 34c “factory custommodified with API transistor outputs and full bantam patch bay with over 80 ins and outs. I recently purchased it off Craig Field in the Blue Mountains, and he imported it from New York where it was originally commissioned to go into a private studio in the Chrysler Building in 1979. I am currently working on showcasing the console’s amazing sound with a cover of Grand Funk Railroad’s Got This Thing on the Move. I’ll be recording it all live with my best clients doing the song some serious justice!”

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Audio Chocolate: (03) 9813 5877 or vwww.audiochocolate.com.au

MOTHER OF SMALL MOOGS $999 | www.moogmusic.com Moog has released the Mother-32; a tabletop semi-modular analogue synthesizer.

Consistent with the Moog heritage, the Mother-32 is an analogue synth housed in an aluminium enclosure with wood sides and adds raw Moog analogue sound, 32-step sequencing with storage for 64 patterns, and 32 audio and control voltage patch points (hence the name) to any electronic or modular setup. It also has a MIDI in for external sequencing and control. It can be used standalone, folded into a Eurorack modular synth, or bolted together with optional racking to connect up to three Mother-32’s in an angled setup. Made in the U.S., it’s an affordable entry into Moog synth products, and its 10-octave oscillator with variable pulse width, and white noise generator, will give you a solid place to start exploring.

NOVATION CIRCUIT $629 | www.novationmusic.com Novation describes Circuit as ‘the grid-based groove box.’ With its 4x8 RGB velocitysensitive grid sequencer, Circuit is designed to be a product that sparks creativity; a ‘start something’ tool to help get over writer’s block. In Novation’s own words, “We wanted something… that gives maximum creative output from minimum input, something that encourages exploration and creates ‘happy accidents’.” Circuit is a standalone, battery-powered box that combines two Novation Nova synth engines with a four-part drum sequencer. It allows you to create entire tunes by combining up to 128 steps of synth and drum patterns, then run it through the built-in effects.

32 slots are available for saving your tunes. You can hook up Circuit to your audio/ synth gear or computer via its line outputs and USB/MIDI ports.

Innovative Music: (03) 9540 0658 or info@innovativemusic.com

ROLAND BOUTIQUE From $479 | www.rolandcorp.com.au Roland has released Boutique — a line of limited-edition modules that recreate some of Roland’s legendary analogue synths. The book-sized units are battery-powered and feature a built-in speaker so you can take that old-school sound with you wherever you choose.

The Boutique family consists of four modules. The JP-08 emulates the Jupiter-8 synth with 36 knobs and sliders from the original Jupiter-8 panel. The JU-06 models the Juno-106 with 23 adjustable parameters and new enhancements such as a faster LFO and continuously variable high-pass filter. The JX-03 module continues the legacy of the dual-oscillator JX-3P synth just as if it was paired with the PG200 controller. The four-voice JX-03 gets all 24 knobs from the PG-200, additional extras like new waveform and cross modulation options, and the distinctive tonal character of the JX-3P. The optional K-25m keyboard unit turns any Boutique module into a compact, all-in-one synth with 25 velocity-sensitive keys.

JP08: $639, JU06/JX03: $479, K25M: $159

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MORE NEWS AT www.audiotechnology.com.au

Roland Australia: (02) 9882 8266 or sales@rolandcorp.com.au

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REVIEW

BOZ DIGITAL LABS +10dB Plug-in We put an original ADR Compex F760x compressor/limiter up against its plug-in version to see if the levee breaks.

Review: Ted Howard

When I open a plug-in and see graphics perfectly matching the look of a piece of vintage hardware my scepticism meter hits the red.

Sure it looks the same, but how does it sound? So given the opportunity to run a comparison between the Compex F760x compressor/limiter and the new Boz Digital Labs +10dB plug-in, I jumped at the chance.

The F760 has long been a favourite of mine during tracking — especially on drum room mics.

It’s got power and wallop but with a clarity in the top end, even in serious gain reduction. Does the plug-in version follow suit?

The session I chose as a testing ground was a recent Ian Moss tracking gig at Rancom St. We have an AKG D109 omni dynamic hanging from the ceiling, affectionately known as the ‘Roof ’ mic.

I ran this through a Telefunken V76m then into an API 550b EQ, followed by the Compex. It’s a big, fat, When The Levee Breaks kinda sound. I also recorded an un-compressed split of the Roof mic post the API EQ for the test.

I opened the un-compressed Roof track and duplicated it in Pro Tools. On one track I patched the Compex F760x via a hardware insert, with settings to perfectly match the recorded track. And AT 14

on the other I opened up the +10dB plug-in. With a little bit of fiddling at the input of the +10dB I had two tracks with the same amount of compression and identical output levels.

The main difference was the limiter function. On the Compex I like to run the compressor (with a fast .25ms attack) and limiter together on the Roof mic — I’m happy to let the close mics handle the drums’ transients and the Roof capture the smeared reflections. The super fast attack of the limiter clips off the peaks and helps minimise that ugly flamming effect that distant room mics can get.

The plug-in needed up to 12dB more input level to trigger the limiter and by then the compressor stage was in super slam mode, quite different from the Compex. The +10dB limiter sounds good (very similar envelope to the Compex) but for me is more useful as a stand-alone limiter rather than a part of an overall compressor/limiter chain.

The other difference was quite subtle; the +10dB had a very slight lack of HF clarity compared to the Compex when gain reduction reached about 8dB and over.

The stand out for me was that the signature power and meat of the original has been replicated.

It really has that sound. After carefully matching

all compressor parameters, I panned the two test tracks left and right (one with the Compex and the other with the +10dB). The two tracks sounded pretty much identical when soloed and when both were open and panned hard it was like listening to a mono source.

I tried replacing the +10dB with various other compressor plug-ins and all of them gave me a left/ right pumping sensation when panned opposite the Compex’d track, even with the attacks and release numbers matching.

Boz Millar has matched the tone and envelope of the F760 compressor really well. In a way he’s improved it; who doesn’t love a wet/dry control for parallel compression? The EQ section, and the way it can be side-chained with the compressor stage is copied from the classic ADR Vocal Stressor; it’s functionally clever and sounds crisp and clean. I also ran a separate lot of tests on the expander stage and it also passed with flying colours.

Overall the +10dB is a really useful tool to add to the arsenal. I can’t wait to try it out next as a vocal channel strip!

Price: US$199 Boz Digital Labs: www.bozdigitallabs.com

STUDIO 192

Built to Create.

The PreSonus Studio 192 is the first USB 3.0 audio interface to deliver exceptional sonic fidelity and flexible connectivity in addition to professional monitoring and mixing controls in a single rackspace. Whether you’re a producer or an audio engineer, an independent band or a sound designer, the Studio 192 was built for you to create.

• Simultaneously stream up to 26 inputs and 32 outputs at 48kHz – or 8 x 14 at 192 kHz. • Take advantage of award-winning StudioLive™ Fat Channel processing on every analog input and the first 8 ADAT inputs with ZERO LATENCY monitoring • Remote control your XMAX™ preamps and create zero latency monitor mixes right from the included Studio One Artist DAW. (3rd-party DAW users can use PreSonus’ advanced UC Surface software to create zero-latency mixes with speaker switching and remote control the Studio 192 preamps via MIDI.) • Set up mixes and control speaker switching from your computer, iPad or touch-enabled PC • Use the Studio 192 as your studio command center with talkback, monitor mixing, speaker switching and two independent headphone outputs • Expand your I/O with up to two DigiMax DP88 8-preamp/converters (Available soon)

Studio 192 is supplied with & fully integrated into the award-winning…

Creative Music Production Environment ARTIST VERSION

Proudly distributed in Australia by

www.linkaudio.com.au AT 15

FEATURE

CURRENT NOSTALGIA Kevin Parker goes ‘old school’ and mixes Currents himself, finding a balance between lo-fi and his own teenage nostalgia.

Story: Mark Davie

Artist: Tame Impala Album: Currents

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“Just got back from tour,” explained Tame Impala frontman, Kevin Parker, settling in for a chat. “Decompressing… no pun intended.”

For anyone that’s followed Parker’s trajectory, compression is no joke. Aside from those catchy psych-pop melodies, Lennon-esque dreamy falsetto, and Dunlop fuzz guitars, the sound of lo-fi, squashed drums was probably the most identifiable part of Tame Impala’s breakthrough album Innerspeaker. And even though the second album Lonerism felt harder to lump in as psych rock, the legacy of Tame Impala continued to be built on that specific kind of nostalgia.

The latest album, Currents, trades on a newer sort of nostalgia, one that — as a teen of the ’90s — resonates more personally for Parker. You can hear it in the hard-ended repetitions of a scratched CD stuck in a loop on lead track Let It Happen, the Boyz II Men R&B drum machine and ’verby claps on Love Paranoia, and the chiming clarity of digital synths throughout.

“I got a few new keyboards. All I had last time was a Roland Juno 106 and a Sequential Circuits Pro One,” said Parker. “I fell in love with those naff ’90s-sounding keyboards. I’ve got a Roland JV1080 synth module you can plug a MIDI keyboard into.

An audiophile would think some of the patches are the cheapest, plasticky sounds. But for me they’re so romantically nostalgic. Because they’re the sounds I remember from when I was growing up in the ’90s.

The sounds that remind me of something I heard on the radio in the car.

“That’s the truest kind of nostalgia I can find.

Those plasticky sounds are far more nostalgic and hit a deeper spot for me than hearing a vintage Fender twin. Even though they’re sort of cheap sounds, they sound deeply fulfilling. There’s a lot of that on the album, glistening FM synth electro Rhodes, like a digital clav.”

It’s a tough break for those young producers who’ve toiled over their Tame Impala emulations: Tea towels on kits, saving up for Juno 106s on eBay, and asking forums whether a dbx 160A would substitute for a dbx 160VU. Now they’re going to have to contend with an inflated market for Roland JV1080s.

Parker is the modern home studio poster boy.

The ideal for a lot of bedroom producers who like the idea of rock ’n’ roll, but don’t necessarily want to deal with band politics when it comes to writing songs. Just like those ‘lucky’ electronic producers, he justified the case for aspiring rockers to forgo the usual band-in-a-studio route, and play everything into a DAW themselves. The demo-in-a-day deal market has shrunk with the rise of a growing contentedness to wile away at the process rather than feel forced to produce in a short window.

Of course, not everyone is Kevin Parker. In fact, no one else is. But it hasn’t stopped leagues of musicians trying to emulate his every sound, image, and pedal board purchase.

IN-HOUSE MIX

While flying completely solo looks nice on paper, for this generation of psych rock-acquainted bedroom producers, one name (outside their own) they’d be happy to see on the mix credit is Dave Fridmann’s.

Fridmann is the hero of indie psych-rock mixing; fearlessly experimental and not afraid to go for colour. Probably most well-known for producing and mixing The Flaming Lips, he’s also leant psychedelic weight to Mercury Rev, Sparklehorse, MGMT, Mogwai, and so many more. When Parker handed the mix for Tame Impala’s second album, Lonerism, to Fridmann, that combo became the benchmark for this crowd.

For his latest album, Currents, Parker decided to take back the reins and do the whole thing himself; play, record, and mix at home. He’s mixed his own music before, and done the same for Pond and Melody’s Echo Chamber, but taking over after such success with Fridmann must have required a newfound confidence. “It wasn’t even really confidence,” said Parker. “It was more like a leap of faith. I just wanted to see if I could do it myself. Dave is an amazing mix engineer. Even as I was doing it myself I found that I was pretending to be him.

“Because of the sound and the way the songs were coming together, it felt like an altogether different way of mixing anyway. I was combining things usually done in the electronic world — like looping whole sections of the mix while leaving others unlooped on top, and filtering whole sections — but at the same time trying to get the drums to sound classic, awesome, and Dave Fridmann-y.”

Fridmann-y is a difficult adjective to describe, says Parker. “Naturally it’s all those nuances you can’t properly describe that he does so well. If you could put it into words, that would mean you could probably copy it really easily. IT’S THE WAY HIS SOUND IS SO CRISPY AND COOKED, SO DEEP-FRIED, BUT AT THE SAME TIME, SO PUNCHY, AND DOESN’T SACRIFICE THE IMPACT THE MIX HAS. THAT’S THE ELUSIVE PARADOX THAT HE’S NAILED.

The drums can be absolutely swampy and sizzling, yet at the same time hit you right in the chest.

“People generally pass that off as an easily attainable thing, just drive it through a preamp and slap a limiter on it, you know. But it’s so much more elusive that that. You only realise how much of an artform that is when you try to do it yourself.

“I’ve got my own experience of mixing that I’ve been getting better and better at over the years.

Which, for me, is the most important thing. I’ve got my mixing style, my drum sounds, and effects I use.

It’s mainly instinct that I’m going on. I just have to hope that it comes out sounding listenable.”

“

They’re the sounds I remember from when I was growing up in the ’90s.

That’s the truest kind of nostalgia I can find

”

A NEW PHASE

When we last talked to Parker, he was revelling in his relative engineering naiveté; recording vocals with a mic wired out of phase, plugging mics into the ‘wrong’ inputs. But this time he stepped up his professional attitude. “That’s not to say it was all the way there,” he laughs. “I still do things that would be laughable to professional recording engineers. To me it justifies the way I do it in a sense. I have a disturbed pride in the fact I don’t do it like everyone else. I don’t do it in Pro Tools. In the end, it doesn’t really matter.

But to me it feels good that I’m a bit rogue.

“Julian, the other drummer in Tame, he played me this album the other day. And I thought he’d gone to an extremely professional studio in France. It sounded amazing. But then he told me he did it all in AT 17

Garageband. I couldn’t believe it. It’s the world we live in these days, there are no rules.”

Taking on the mix was as much about gauging his current ability to pull it off, as it was an excuse to get some new gear. Gear only a professional would use. “It was a good excuse to buy stuff I could never justify getting, unless it was something as important as mixing my own album. It sounds selfish. I’m not a professional studio owner, who lives off mixing people’s albums. So I’ve never really bought gear that is just involved in mixing.”

The main adjustment to his mixing routine has been an analogue summing setup that takes that task out of Ableton Live’s hands and adds a bit of colour to the stereo bus. He bought two 16-channel Lynx Aurora converters that feed a pair of passive 16-channel RMS216 Folcrom passive summing mixers. The sum of the 16 stereo channels hits a pair of Neve 1073DPA preamps first to “crunch the mix up” and make up gain, then an SPL Vitalizer, the one piece he’s had for a while and is “just like a glorified EQ that adds a bit of artificial valve quality.” The last thing strapped across the master bus is a Manley Vari-Mu compressor before it gets fed back into the DAW to print. The gear doesn’t see any use other than for mixing, so it allows Parker to leave it set up and return to any mix at will. “I’ve always loved mixing with Ableton,” said Parker. “But I love the sound of a mix being crunched together in analogue. I love the sound of a desk, but I hate not being able to recall any time I want. I flit between songs, sometimes every few minutes. It goes completely against my workflow to have to set up for one song by going around the room and setting EQ knobs on a desk and adjusting faders to how the mix was last time. It’s against how I’ve grown up working.”

It’s a big step up from “just whatever plug-in in Ableton” he used to place over his master bus. He’s always had outboard channel compressors, but never pressed them into service over his whole mix.

That’s no knock on Ableton though, Parker is still a dedicated Live user. “It’s ultimately just what you’re used to. Everyone barracks for their favourite, like a football team. But for me, I find Live so expressive. The whole point of it is that you can link anything to anything, and the automation is amazing. When I’ve been standing over someone’s shoulder watching them use Pro Tools to automate stuff, it seems so laborious. It doesn’t want you to automate. It doesn’t want you to change things that are so easy to change on Ableton.

“I make our live shows on Ableton as well, which demonstrates how versatile it is. The basis of it is just so flexible, and I can move almost as quick as my brain is moving.”

HYBRID DRUMS

Parker recorded Currents in his home studio, where he has two rooms joined; one acting as a control room, the other houses his ‘mongrel of a kit’.

“Parts of it are from my old ’60s Ludwig I used on Lonerism,” said Parker. “The kick is a vintage Tama kick I’ve had forever.” As for mics this time, Parker wasn’t that fussed about sticking to a set routine. “It was a bit of a range this time. Between several mics AT 18

to just one, it really just comes down to how set up the studio is at the time I’m inspired. If I’m inspired to record some drums and there’s only one mic there, then I’ll just say, ‘f**k it.’ Just record it with one mic and go with it. I’m not precious. I used a Shure SM7 on a lot of stuff, for drums and vocals.

“A lot of the drums were a cross between the real kit, a sampler on Ableton with something deep and woofy for the kick, and some drum machine sounds. I’m in love with the Sequential Circuits Drum Tracks drum machine, it’s my favourite drum machine. I recorded a lot of claps myself and used them as a set sample. I just recorded a bunch in a row, with a different hand position each time and put them all together at once, spaced slightly apart. I try and get the best combination so it sounds the most feel good, ‘party times’.”

For guitar recording, Parker uses his go-to guitar signal chain: “It’s the same pedal chain I’ve had for ages; reverb, a bit of compression, a bit of overdrive.

I have this Seymour Duncan rack preamp that I love. They didn’t make very many, and it’s not even very good quality, but for some reason, I can’t find anything that’s got the same kind of sizzle to it. I DI all guitars… I haven’t used an amp in years.”

There are some heavy riffs on the record, especially the one that pounds out the choruses in the otherwise dream poppy Eventually. But, rather than layering lots of parts, Parker says, “It’s just a guitar riff, bass riff, and a synth that’s escalating in pitch to add some tension. I find that when you try and make something heavy, the less elements you have in it, the bigger it sounds. If you try and make a heavy riff with seven guitars, three different types of synth, a bass synth, and a bass guitar, it really just muddies it up and loses impact.”

TRANSITIONING OUT OF LO-FI

These days, Parker is using lo-fi elements as a transition effect. Similar to how mix engineers often make the chorus wider, he’ll just dial up the nostalgia in the intros and verses to lull your ears, before hitting them with the power of a fullspectrum sound. It happens a few times across the

record; band-passing the drums and guitar intro on The Less I Know Better, and turning the first half of Disciples into a radio pop single from the ’50s. Parker: “For the static on Disciples, I tuned my car stereo to an AM station that wasn’t an actual channel, recorded the static and mixed that in.

Someone told me that AM didn’t have anything past 6kHz. So when I mixed it, I JUST LOPPED EVERYTHING PAST THAT POINT TO MAKE THE WHOLE TRACK SOUND LIKE IT WAS COMING OUT OF AN AM RADIO. I used

a multi-band compressor and followed the middle band, so it had that compressed, boxed-in sound.

When you turn it off, the song suddenly opens up.

I love switching between the two, because your ears adjust to that set of spectrum. It’s like, ‘ok, this is what I’m listening to.’ Your ears forget about the sub and the super top. And all of a sudden, when they kick in, it’s like this sensory, pleasurable thing.

“I’ve always tried to make things hi-fi, but I just haven’t had the ability. At the same time I’ve always loved lo-fi sounds. To me, they’re always the most instantly cinematic. Anything lo-fi, wobbly and crusty instantly transports you. In the past, I’ve used that effect, but it was the whole sound. But this time I’ve been embracing different kinds of textures.

“Not just lo-fi sounds, but juxtaposing sounds that are super clean and super crisp. It makes it sound more like I’ve sampled something and put it in there. I was obsessed with that idea, in the way that hip hop has this really high-quality vocal, and maybe a high-quality drum machine, then the guitar that comes in is totally crusty and boxy sounding. I love that kind of juxtaposition of sound quality.”

Already a master of nostalgia, this new sort Parker has added to his sonic repertoire perfectly complements the eclectic nature of his new material. While it was wrong to pigeonhole Lonerism as just psych rock, Currents is completely new ground for Tame Impala that feels vaguely familiar.

“

I have a disturbed pride in the fact I don’t do it like everyone else.

To me it feels good that I’m a bit rogue

”

AT 19

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FEATURE

HITS THE SPOT The planets align for Jamie xx & Tom Elmhirst, who play the percentages of digital and analogue to lend gravity to In Colour.

Story: Mark Davie

Artist: Jamie xx Album: In Colour

AT 22

“There’s no fat with his music,” is how mixer Tom Elmhirst would sum up the In Colour multi-tracks Jamie Smith (aka Jamie xx from The xx) handed to him to mix. “I’m not going to lie,” he said multiple times. “He’s done alright without guys like me around. It was 97 to 98% there. He’s so above average, and I mean that as a massive compliment. What he does on a laptop and a pair of headphones is not available to all of us.”

Elmhirst first heard the now 26-year old Jamie, when The xx were performing one of their first shows, back in the day when they were still a fourpiece — “18 and shitting themselves,” remembered Elmhirst. “They hid behind loads of dry ice. I’ll never forget running over to Richard during their set saying, ‘Why the f**k have you not played this to me?’” That’s Richard Russell, head of XL records and an old friend of Elmhirst’s. “It was brilliant even then!”

While Elmhirst hasn’t gravitated towards mixing a lot of club music over his career, he practically cut his teeth on the genre. He started out at Sarm Studios in the Whitechapel area of London, where Queen recorded Bohemian Rhapsody. And in the late ’80s, Ministry of Sound, down on the corner of Elephant and Castle, was dominating the nightclub scene. “All the American guys would play Ministry and do remixes at Sarm East while they were there for the weekend,” said Elmhirst. “They’d start quite late in the day, do a remix, then go straight to Ministry. My very first sessions in the studio were with these amazing American house guys — Masters at Work, Derrick May, Kevin Saunderson — who came because we had custom Eastlake speakers, an SSL, and a really special sounding studio. We’d do remixes on an MPC, E-Mu SP1200, and a multi-track. It was amazing.”

Growing up in Ladbroke Grove, he also experienced a lot of West Indian culture that permeated London’s music scene, and the prominence of steel pan in Jamie’s music sends Elmhirst right back to the Notting Hill Carnival.

“The steel pan was all around us, but what he did with it was very different,” reckons Elmhirst.

“Jamie’s a melodist. He stumbled across the steel pan and did something quite different with it. He took it and kind of made it his own, in the way Ron Burgundy made the jazz flute his own, or like Stevie Wonder and a clav.”

NOT THAT NEW HERE

It was actually Jamie’s work with Gil Scott-Heron — remixing the album I’m New Here Heron had recorded with Richard Russell into We’re New Here — that really showcased his production chops. Up until that point, he’d released a few remixes, and everyone had an inkling that much of The xx’s powerfully minimalist execution had a lot to do with Jamie. But being able to compare an entire album’s worth of original music with his remixes quantified the extent of his influence. It was a lot.

He’s now a sought after producer, lending a reverberant house piano and signature xx guitar line to Drake’s Take Care, and bringing an enchanting filtered ambience to his co-production of Alicia Keys’ When It’s All Over. He could have

easily cashed in on those credits, but instead shifted his efforts into finishing In Colour. Some of the tracks, The Rest Is Noise in particular, were in the works for a long time and really needed to get over the line. “I worked on that song for about five years,” explained Jamie. “Which is why it goes on such a journey.” It’s a reason so many great records have come out of the XL stable, as Elmhirst explains: “Richard likes people. So when he likes you, he’s going to trust you. He’s not dealing with the weaklings; they’ve got a point, and he just lets them get on with it. He thinks they’re the best taste makers for themselves. That’s really empowering for an artist.”

For Jamie, it was about having the space to finish the album on his own terms, without that pressure of other jobs. “They’re not really worrying about trying to get me to do stuff,” he said of XL. “It’s more trying to finish stuff.”

HUMAN TOUCH

Jamie’s studio in London is constantly evolving to suit his needs. “I did have a console for a while but I prefer to use preamps straight into the Apogee Symphony.” It’s not devoid of analogue gear, the “vocals for Loud Places and See Saw were recorded at my place in London,” he said. “I always use a Neumann U87 with Neve preamps.” He has a handful of compressors he uses to manipulate sounds out of the box, and among some effects units, a hi-fi spring reverb add-on unit from the ’70s. “It’s a separate unit you can rewire to use as an outboard piece,” he explained. “It helps give a little bit of life to certain things.” It’s not about hanging on to any one piece of gear but gathering the things that inspire his creative juices. Sometimes that means going back to the most basic of instruments.

“I find it easiest to play the piano, because that’s what I know how to play best,” he said. “I just play until I make something that sounds new to me, that’s different, and has a harmonic to it that gets you every time.”

In Colour eschews the sometimes unholy cleanliness of electronic records for a more organic sound, tailoring the top end and emphasising the lows. Music that lifts its inspiration from vintage samples, even amongst recent proponents of Garage House, can often treat the process as a ‘one and done’ affair. Either bedding samples into a disparate ambience that leaves them poking out to carry ‘the vibe’, or burying them into the mix as token colour. Jamie does neither. By ensuring his productions are organic in their own right, he gives samples a new place to exist that feels neither foreign nor nostalgic. “Though a lot of the stuff I use is electronic, I didn’t want it to sound too clean; I want it to sound human, and almost live in a way,” said Jamie. “It’s a little about how I’m treating sound, but it’s also about making things less rigid than they would be if I’d just left them as they were. The synth on The Rest Is Noise has been side-chained and re-side-chained several times through different compressors at different levels.

It’s side-chained to the kick at several different points, and it creates these little peaks that change throughout the track, because most of the kicks

He was quick to understand how he wanted electronic music to sound — and that was, it should sound performed

AT 23

weren’t quite on. It makes it sound really different to how it would sound if it was just done in the box, compared to using an actual compressor.

“Things happen quite naturally. Most stuff I’m happy with just flows out and ends up sounding good because I’m happy.”

He offers up Gosh as an example of working on a track till he finds the right feel. “It originally started out at 150bpm, but ended up sounding like a jungle track and didn’t have anything special about it.

Then I used Varispeed on Logic, and slowed it down to 143bpm. It gave the whole tune a lot more space, and felt very analogue, but was still just in the box.”

Elmhirst cottoned on to Jamie’s organic style from that first The xx performance: “He curates the sound, and it’s performed. He plays a drum machine into a sequencer, he doesn’t let the sequencer govern him. From a production standpoint, he was quick to understand how he wanted electronic music to sound — and that was, it should sound performed.”

IN THE PERCENTAGES

This was the first time Jamie has turned his album over to a top-flight mixer with such a wealth of analogue gear. Similarly new for Elmhirst, in a career that’s spanned a couple of decades, the In Colour session was one of a handful where he had the artist in attendance for the entire duration. He doesn’t think mixing is a “spectator sport”, preferring to just work with his assistant, Joe Visciano. “I find many artists have a better perspective when they come in later in the day and the mix is quite close.” But having Jamie in the room wasn’t exactly distracting, Elmhirst pointed out: “He’s a man of few words. So when we did talk, it was quite quick and to the point. He’s super quick to go, ‘nah, I don’t like that.’ We mixed that whole album in six or seven days.”

When Elmhirst spread the entire mix across his Neve VR console at Electric Lady Studio C, he was searching for that last few percent to bring to the songs. “I added a little bit of dimension and width, the depth of field which I don’t think you get from a laptop.” said Elmhirst. “And some analogue processing.” It’s Elmhirst’s ability to be so frank about his involvement level that makes him such a powerful ally. While he reckons his contributions to a mix might typically add 30-40%, he’s not automatically looking to stamp his imprint onto a record if it doesn’t need it. Even the small percentages he did bring were made to order. “It was a case of, ‘What do you want to do? Go wider, deeper?’,” said Elmhirst. “I think he enjoyed sitting at the back of a room and listening.” Every now and again, Jamie would ask to listen to it on headphones, recalled Elmhirst, which was usually a sign he was happy with the mix. “Usually what works in headphones will work in a club,” reckons Jamie.

It was that dual outcome Elmhirst had to keep in mind: “Jamie wants it to be a) listenable at home or in your car, but b) also wants it to pound in a club.”

To meet that second criteria, he often deferred to Jamie’s judgement. “Jamie is much more in tune with club culture,” said Elmhirst. “So sometimes the AT 24

vocals aren’t going to be the most important thing.

Loud Places is not that loud if you listen to it. She’s not on top, it’s very much part of the fabric of the track. Not where I’d put an Adele vocal. I’d want to turn her up, but Jamie would be, ‘nah, nah, she can be inside it.’ That’s the fun process of making records. You allow everyone to do their bit, and you collaborate, and it all works out in the end.”

Halfway through mixing the album, they hired a club in Brooklyn and invited a bunch of friends down so Jamie could play the record in amongst a DJ set. “It’s such an important part of mixing club music,” reckons Elmhirst. “To road test it right there and then. It’s like being a chef. It’s very quick from A to B. If something doesn’t feel right, you know straight away. Jamie did come back with some changes after that.”

He took it and kind of made it his own, in the way Ron Burgundy made the jazz flute his own, or like Stevie Wonder and a clav

It’s about finding “that communal moment, like being at church, when certain things make sense and you get to share them with a bunch of other people,” said Elmhirst.

CONDENSING THE WETNESS

The lack of any fat in the multi-tracks was the best starting point for Elmhirst’s mix. After all, he said, “all we’re doing when we’re mixing records is trying to condense an idea. If you start with 40 tracks, it goes to 10, then left/right stereo. It’s a constant state of reduction.”

Some of the tracks were sparse in their own right. Stranger In A Room, for instance, was just two stereo synth/guitar tracks and a vocal.

Jamie says he tries to “use as few instruments as possible to make the sound I want to make.”

His main synth, an Oberheim 8-voice, fits that bill perfectly. “It’s like eight different synths just piled up on top of each other, and takes forever to make something on,” he says. The piling on of voices into the one patch making it “feel like a don’t really need any other synth.”

Although it wouldn’t seem like it, songs like Stranger In A Room are probably where Elmhirst can have his biggest impact. Vocals are really important to him. “I did a lot on that in terms of

keeping Olly’s vocal in-your-face. The vocal sounds like he’s sort of talking to you in your ear. But there’s actually a whole bunch of stuff going around behind him, it’s just not in the forefront.

“There might be seven or eight effects on the vocal, but you can’t pinpoint one. You can’t go, it’s Preset 44 on a Lexicon PCM80, because I’d rather shoot myself in the head. I might use seven reverbs on one vocal, but they’re all doing very different things, and all EQ’d differently. I’m not a huge fan of high end on a vocal reverb, I don’t like when an ‘s’ goes ‘tsss’. In the old days we used to use an SPX90. And there was a preset, I think it was 23, called Pitch Change C. It was like having NS10s in the studio, you had to use it on the vocal. But I’d rather use eight effects incrementally working together.

“I use quite a lot of Lexicon Delta-T, which was a box used to offset speaker stacks in stadiums.

The maximum delay time is 122ms, it’s hybrid digital/analogue, but it’s a very warm delay. I use a slight offset left and right delay, quite short; 90ms on one side, and 100ms on the other. I also use a bit of chorus, I really like the Roland CE-1 chorus pedal that tragic guitarists used in the ’70s. But you couldn’t necessarily pick it if you sat there listening to the vocals.”

Jamie also uses other effects to give a sense of depth. Particularly, he likes using “a phaser effect to place something in the mix. But using it to put it in one place, rather than having it phasing and moving about.”

Every little nuance contributes to the organic sense of depth and width, even that last two or three per cent, said Elmhirst: “If everything’s loud all the time, there’s no dynamics. If everything’s dry, there’s no background. If everything’s wet, there’s no foreground. That’s the interesting thing about mixing records.

“Jamie likes quite short rooms. I like when the vocal is quite present and in-your-face, but there can be things behind it. It’s all about placement, where you put instruments and vocals to create space. The way I work, which is still on an analogue console, I have sends on the vocals. And I can alter in an arrangement — between verse and chorus — how much is wet, dry, delayed, reverb.

“It’s the same in film, where some directors will use music or visual prompts to provoke a reaction, while some will use the inverse. Silence where there should be grand orchestration, and close shots in a big space.”

COSMIC COINCIDENCE

The connection between film and mixing isn’t just philosophical for Elmhirst. He always plays visuals on the big screen while he’s mixing. When he mixed the Arcade Fire album, it was weeks straight of Kill Bill; “there’s a balletic quality to martial arts that lends itself to the precision of making records.”

Other times, he’ll have nature reels like Planet Earth on loop in the background.

“To me, records should be very dimensional like that,” he said. “It’s not just left and right, it’s like film — foreground, background. It’s focus, vertical dimension, and width. Why can’t they

I just play until I make something that sounds new to me, that’s different, and has a harmonic to it that gets you every time

AT 25

STRANGER IN A ROOM Tom Elmhirst’s Assistant Engineer, Joe Visciano, gave AT the rundown on how Tom implemented effects on Oliver Sim’s lead vocal in Stranger In A Room: “The lead vocal is always on channel 25, because it’s the first channel to the right of the centre section.

So when Tom is working the vocal, he’s most central between the speakers. He sends the vocal via bus output, returning to three channels (but off the bus). From there, he uses the auxes to send to reverbs (usually Lexicon 480, EMT 140, Dverb or TL spring), a delay (Lexicon PCM 42 or AMS DMX), and a delay into reverb (AMS DMX into an Eventide H3000) each on their respective channels. This allows highly flexible and immediate effects options for the vocal during a mix.

“The mix bus is always the same; Chandler Curve Bender into a Manley Vari Mu on the mix insert, and the CraneSong HEDD as the master AD converter.”

Before sending the lead vocal out to the effects, Tom treated it with some standard de-essing, before adding some harmonic juice with the Massey Tape Head and Soundtoys Decapitator plug-ins, then finally using another de-esser at the end of the chain to tame the really high end.

You can also see how Tom rolled off the Synth Hi track above 4kHz, in keeping with Jamie’s organic sound, and used the UAD Fairchild 660 plug-in to compress the simple guitar line. The entire mix is simple, and a lesson in restraint, letting the vocal take the lead.

inform each other?”

Despite painting himself out of the picture. One thing Elmhirst will take credit for is the inspiration for Gosh’s music video. Which is an epic animation directed by digital artist Erik Wernquist about a future terra-formed planet that looks a lot like Earth on its far side.

In perfect cosmic alignment, while he was mixing In Colour, he had Alfonso Cuarón’s Gravity playing for the entire time. So enthralled is he by director Cuarón’s recreation of microgravity and the Light Box/robot-controlled camera combination used to technically execute the film, when I talked to him months after Jamie had left, it was still onscreen at Electric Lady.

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“The film was groundbreaking on a technical level,” enthused Elmhirst. “It makes creating a record look so simple. They built a light chamber with LED panels, and then put this rocking horse thing in the middle so the actors could rotate, while robots that are use in car manufacturing manoeuvered the cameras.

“It relates to mixing, because in mixing, you have such a long learning curve, to understand compression, limiting, EQ, balance. And then you have to forget it all. It’s like learning to ride a bike. Once you know how to ride a bike, you don’t remember how you learnt, you just ride it.

“I spent 10 years learning how to do this, and now I just try and concentrate on song, melody,

lyric, artist. Going back to Gravity, what’s brilliant, is it’s so staggeringly technical, but when you watch it, it’s just a story about human evolution, pain, loss, life, love — all the standard jazz. For it to be technical, but emotional and emotive, and a story about human pain, love and loss is brilliant.

“I think In Colour achieves a similar objective.

But seen through Jamie’s eyes and his experiences with London club culture.

“There’s a galactic, yet universal quality to that record — it’s very soulful.”

SX-SERIES

“Wonderfully detailed, with an exceptionally transparent midrange.” Paul White, Sound On Sound on S3X-V “ADAM is pushing the standard for transparency in a monitor system to new levels.”

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“We started off with a classical string quartet recording and let’s just say right away that the S2Xs floored us. Several adjectives spring to mind: wide open, transparent, effortless, breathtaking realistic, loud, dynamic, involving. [...] These monitors are up there with the best we I’ve tried so far.”

Huw Price, Music Tech

“The S3X-H convinces all along the line. ADAMs new member is all but just an improvement of a successful monitor. With a new loudspeaker design and painstaking tuning it is a first-class control tool in any studio. From now on, the monitor reference in the Professional Audio Studio is called S3X-H.” Raphael Hornbruch, Professional Audio

For dealer listing, please visit www.federalaudio.com.au/dealers AT 27

TUTORIAL

HOW DO YOU MEASURE UP? PART 2:

OFF-AXIS IS YOUR ALLY Sometimes it’s not all right in front of you. A speaker’s off-axis and power response has more effect on your measurements than you might think.

Tutorial: Ewan McDonald

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Your mum hands you a sack of second-hand Lego and tells you that buried amongst the green boards, rubber wheels and decapitated heads are all the pieces to build a pirate ship. All you have to do, she figures, is separate the ship bits from the miscellaneous pieces and you’re good to go. “Oh, and by the way,” she trails off as she closes the door, “I threw the box out years ago.” Along with any idea of what the ship looked like and instructions for how it’s supposed to go together. Thanks mum!

You set down your Han Solo figurine and rumble through the pieces trying to pick out ship-looking bits. Your best guess is that it’s majority brown but other than that, if it isn’t a sail and doesn’t have ‘pirate’ written on it, you’re just speculating. After half a day you end up with a mountain of ‘maybes’ and a handful of ‘definitely nots’, but aren’t any closer to seeing the ship through bricks.

Much like that sack of Lego, when you measure in a complex acoustic environment the response of the loudspeaker is jumbled in with the effects of the surrounding environment, creating a puzzle that can be very difficult to solve.

BUILDING BLOCKS

To get the most out of your measurement system, you need to be able to break down and analyse what you see. In the case of real-world loudspeaker measurements, it’s about being able to separate the response of the loudspeaker system from the contribution of the room and the surrounding acoustic environment. If you know what the loudspeaker looks like in a reflection-free environment, it’s much easier to determine the acoustic environment’s effects. It also simplifies the process of deciding which parts of the trace should be equalised and which should be left alone.

Now if this seems daunting, don’t worry, I was hopeless at measuring PA systems in reverberant environments for a long time; often just turning the computer off and doing it the old-fashioned way. I just couldn’t see how the measurements correlated to anything I was hearing. But once I had a detailed look at speakers in a controlled environment, things became much easier to decipher.

IT’S PLANE, ON REFLECTION

The previous article (Part I, in Issue 107) demonstrated it’s actually fairly easy to spot a discrete ground reflection in your measurement trace (both speaker and mic on stands) when compared to a ground plane measurement (speaker and mic on ground). But when you take that loudspeaker indoors you will have plenty of other things to deal with. Apart from the loudspeaker itself, and interactions between drivers within that loudspeaker, you have interactions between multiple loudspeakers, coupling between the loudspeakers and surrounding areas, reflections, reverberation, resonances… the list goes on.

To begin solving this puzzle we need to know our speaker more intimately than what we can garner from a simple ground plane measurement.

First, we’ll revisit the on-axis response of the loudspeaker in a little more detail. Believe it or not, this is actually the least important measurement, but usually the one people care about most.

Next we’ll look at the off-axis response of the loudspeaker; what the loudspeaker is doing in all directions, not just directly in front of it. It will give us some clues about how the speaker reacts when measured at different points in the room, how it behaves in a reverberant environment and how it interacts with other speakers in an array.

Much like the first article, having a better understanding of this doesn’t just help you read measurements, it improves your practical understanding of audio theory.

CUT DOWN THE MIDDLE

Let’s start with the on-axis response. While more loudspeakers are shipping with ‘flatter’ responses than in the past, without measuring them yourself, you’ll never know the actual response. For this experiment to be truly effective, you’ll want to measure a loudspeaker you’ll actually use in the field. If you can’t get hold of one, don’t despair, you’ll still learn a thing or two with any speaker.

Some of the world’s most popular loudspeaker systems have terrible off-axis responses, so you cannot take it for granted

A bit of a recap from the last article. First, get away from reflections. Outdoor measurements away from obstacles will usually work best, provided wind is not a factor. If inside, you’ll need a very large space — you’re living room will be too small. Second important step, label your measurements so you know what you’re comparing later on.

Now, take the on-axis measurement using the ground plane method outlined in the last article.

Once you’ve got a nice reflection-free ground plane measurement, you want to capture some traces at a measurement distance you would generally use when measuring that speaker in the field. It will give you a practical reference for later use.

Make sure you label those measurements with the distance from the speaker and store them.

Now look at that measurement. Is it flat? Does it have a smiley face curve, some big dips or peaks, or a large boost in the high frequencies? Where do the low frequencies and the high frequencies roll off?

Get a good idea of what the speaker is doing.

THE MOST COMMON MISTAKE AFTER REALISING THE SPEAKER DOESN’T HAVE A RULER-FLAT RESPONSE STRAIGHT OUT OF THE BOX, IS TO THINK THE MANUFACTURER GOT IT WRONG AND START REBUILDING THE LOUDSPEAKER PRESET TO MAKE IT LOOK BETTER. If that’s you, then stop right

there! On-axis response is only one very small piece of the puzzle. Having helped build loudspeaker presets, I can tell you it’s no trivial process. Unless you understand the complex world of loudspeaker and crossover design, it’s best to leave it alone. That being said, there are plenty of speakers out there that aren’t perfect. In a case where you consistently notice something in the field (e.g. you consistently put a filter at 3kHz), then you may be able to use those measurements to help you refine that filter.

OFF-AXIS BEHAVIOUR

Next, you’ll need to understand how the speaker behaves off-axis, which is much more important than the on-axis response when you’re measuring in a reverberant environment. We’ll start by doing some polar measurements and then discuss something very important called power response.

Recognising dips in polar responses (both vertically and horizontally) can be really handy when analysing traces in the real world.

Manufacturers usually supply polar responses, which are a great reference provided the data is accurate and you know how to read them. Some of the world’s most popular loudspeaker systems have terrible off-axis responses, so you cannot take it for granted. You’ll want to have your own accurate measurement that you can pull up to compare with real-time measurements.

Polar measurements are usually done in a couple of different ways; by rotating the microphone around the loudspeaker, or keeping the microphone stationary and rotating the loudspeaker on the spot. The second method requires less space, so we’ll go with that.

First, start with the same setup as your on-axis measurements. YOU NEED TO BE IN THE ‘FAR FIELD’ FOR POLAR MEASUREMENTS, SO GET AS FAR AWAY FROM THE CABINET AS YOU CAN. At least four metres away is

great, but the distance will be cabinet size- and space-dependent. If the cabinet is symmetrical in the plane you’re working in (either horizontal or vertical), you’ll only need to measure one half.

If you want to measure the vertical polar response, simply lay the cabinet on its side (making sure it’s pointing straight out, not lying on an angle) and re-measure. Every 10 degrees should be fine, try less if you’re really keen. Most manufacturer’s data is measured in five degree increments but the increments don’t really matter. The aim is to give you a reference of how the speaker behaves off-axis. If the speaker response exhibits a big dip at a certain point, you should expect to see that when you measure from the same angle in the field.

Next time you’re measuring that loudspeaker, pull up those measurements you’ve stored and labeled and see if they relate to what you’re seeing.

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Now when you see the dip in the field you’ll know it’s an artefact of the speaker/crossover design rather than a reflection, or something else, causing it. Let’s look at an example. Here’s a two-way speaker measured in the vertical plane, both onaxis and approximately 15-degrees off-axis.

Note the big dip that appears around 2kHz, due to the misalignment of the low and high driver. This will show up in varying degrees when measuring at the same angle in the field.

Now you know it’s there and can identify it in a measurement, you’ll be less inclined to reach for the EQ to try and fix it.

TIP: GOOD ANGLE Since you’re probably not going to have a programmable turntable, to rotate your loudspeaker to perform the polar measurements, you’ll need a protractor to get accurate angles. If you don’t have a novelty oversized protractor on hand, just search Google images for one, enlarge it as big as it will go and hit print. Since your speaker will be sitting on top, extend your angles with chalk or gaffer tape so you can see them and rotate the speaker accordingly.

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Picture your loudspeaker with an imaginary sphere around it. If you were to measure the frequency response at every point on that sphere and then calculate the sum of all those measurements, that would give us the power response.

If you put that loudspeaker in a highly reverberant environment and take measurements in the reverberant field — i.e. beyond the critical distance: the point where the level of the loudspeaker’s direct sound equals the sound from the reverberant field — the power response curve would dominate the measurements, not the on-axis direct response. This is because the reverberant field is being excited by sound travelling in all directions from the loudspeaker, not just what comes out the front.

This is why the power response is so important.

YOU CAN HAVE A PERFECTLY FLAT ON-AXIS RESPONSE, BUT IF THE POWER RESPONSE IS NOT SMOOTH — AND YOU’RE IN A REVERBERANT ENVIRONMENT — THOSE BUMPS AND DIPS WILL GENERALLY SHOW UP IN YOUR MEASUREMENTS AND YOU’LL HEAR THEM COLOUR THE SOUND OF YOUR PA SYSTEM. As you see and hear these, if you start

to correct them, what you’ll actually be doing is starting to degrade the on-axis response and you’re left with a transfer function that may look flat but will certainly not sound that way.

SPHERE OF INFLUENCE

FEEL THE POWER

Now let’s talk about Power Response — something you’ll probably never actually measure, but you’ll need to understand the concept. When we measure in a reverberant environment we see both the direct sound (relative to the angle we’re listening at), plus the energy in the reverberant field. The energy in the reverberant field — which we call the power response — is made up of the sum of all the radiated sound from the loudspeaker in all directions.

AT 30

So what does the power response look like? Sound becomes more omni-directional as wavelength increases, relative to the size of the sound source.

So if we’re measuring on an imaginary sphere surrounding the loudspeaker, the lower frequencies will generally show up in all the measurements, while the high frequencies will tend to show up more in front of the loudspeaker. This results in a power response that is tilted downwards, from low to high frequencies, but with various dips and peaks, usually based on interactions between the

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drivers within the cabinet.

If you measure a PA in a highly reverberant space, your measurement system does a good job of giving you a fairly clear picture in the high frequencies. But it cannot remove all the additional low frequency energy from the reverberant field.

You’re not just measuring the on-axis response but a combination of the on-axis response and the power response.

Since the power response is supposed to be tilted downwards and dominates the measurement as you move further into the reverberant field — if you equalise it flat again, you’ll actually be tilting the on-axis response upwards from the low to the high frequencies, resulting in a harsher sounding system.

Also, if the power response is not a smooth slope, but has some fairly large peaks, those peaks will generally show up in a reverberant field measurement. As you remove them, you will again adversely affect the on-axis response to compensate. Now some of you may ask, “but if the energy in the reverberant field is contributing to the response, shouldn’t flattening it off give us a more ‘linear’ room response?” The problem occurs when we actually have to listen to the PA system.

Our ears deal quite differently with the initial direct arriving energy from the loudspeaker and the late arriving, time-smeared energy from the reverberant field.

So you’re probably wondering where you can get your hands on the power response for your loudspeaker of choice. Well, unfortunately, you’re probably out of luck. Most loudspeaker companies don’t even calculate it let alone release it, choosing to work from polar responses instead. So while you may not be able to ‘see’ the power response in a graph, you’ll need to understand the concept and realise that it’s just not what comes out the front of the loudspeaker that counts when you’re measuring inside.

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POLAR CAPS

Compare these two graphs: If you’ve never looked at a polar measurement in this form: they display frequency, level in dB and degrees off-axis in the one graph. In the first polar we can see that it’s very omni-directional at 100Hz then becomes more directional as it increases in frequency before becoming wide again just below 500Hz. Above this the coverage narrows before becoming wide again at 1kHz. It narrows again before stabilising about 3kHz. Compare this to the second polar. This is also very omni-directional at 100Hz but quickly stabilises below 500Hz and remains fairly consistent right up to beyond 10kHz. Now let’s look at some frequency response measurements. Both have been normalised so they have a flat on-axis response. We can see that in the first example, these bumps in the polar response around 500Hz, 1kHz and 4kHz can be seen in the frequency response plots off-axis.

If we measured this loudspeaker off-axis and then tried to flatten the trace, we would be adversely affecting a large portion of the room.

You can see the second loudspeaker behaves very similarly throughout the coverage area apart from an off-axis dip just above 1kHz. Any adjustments AT 32

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to this loudspeaker off-axis would make much more predictable results through the rest of the listening area.

Now, imagine in a perfect world that your venue has filled with people and the reverberant energy has diminished significantly. If the on-axis and off-axis responses are similar you will have a system that will only require minimal adjustments to compensate for the change in the reverberant field.

Now you have a better understanding of your loudspeaker, the fun part begins. Take your loudspeaker and put it close to walls, put it in different sized rooms, on the floor, above the floor, put it in dead spaces, reverberant spaces and keep on measuring. Notice how much those responses change. Notice which parts stay the same. Ask yourself — how has the response changed placing it near a wall? Can I see the reflection from the wall in the measurement? Can I see any loading in the low frequencies when placed near the wall or floor?

Can I see the room modes? If I’m in a reverberant room, can I see the additional energy in the lower frequency region? If I move off-axis, do any of those dips correlate with what I saw in my off-axis measurements?

Let’s finish with an example. Look at this trace and ask yourself what you’d do.

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Once I had a detailed look at speakers in a controlled environment, things became much easier to decipher

ABOUT THE AUTHOR Ewan McDonald has measured just about every manner of PA system all over the globe — from boardrooms, dingy pubs, and churchs, to opera houses and stadium PAs. After spending many years as a senior system engineer at Norwest Productions and an Applications Engineer at Adamson Systems Engineering in Canada, Ewan has since settled back in Australia as a System Sales Specialist at Technical Audio Group in Sydney.

In Room Trace Only

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Now take a look at this trace compared to a semi-anechoic response (in blue) and ask yourself the same question.

While it doesn’t tell you what to do next, this does give us a much clearer picture of what part of the trace can be trusted, and which parts are obviously being affected by the room or the position of the microphone. Whatever decision we make next will be better informed than if we didn’t know how the loudspeaker performed outside the room.

As you do this more, you’ll not only end up with a more intimate understanding of your loudspeaker, but you’ll gain a better understanding of the acoustic interactions around you, which is the ultimate goal if you want to start getting the most out of your systems.

Happy measuring!

Special thanks to my good friend Benoit Cabot for helping provide some of the data.

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REGULARS

PC Audio Microsoft Surface — in touch with audio? We give you the lowdown.

Column: Martin Walker

In the last couple of issues I’ve pointed out the perils of buying third party laptops/ tablets for audio, and extolled the virtues of the good ol’ PC desktop for same. However, there is a third path, and that’s buying a tablet direct from Microsoft. As I explained in AT108, generally the big problem with laptop/tablets is that you don’t know what array of components they contain, or whether any of these (and their associated software drivers) will interrupt your audio stream and cause clicks and pops in your recordings, without the ability to disable them and cure such issues. Well, now that Microsoft has its own Surface range of tablets, we finally find ourselves on level pegging with our Mac brethren. Because just like them we are being offered a device with a fixed and well-known specification, with hopefully no unpleasant surprises in the audio department.

BELOW THE SURFACE

Although there is already speculation about Microsoft’s Surface Pro 4 being released in late 2015 or early 2016, the current models to look at are the Surface Pro 3 range and its cut-down cousin the Surface 3. The significantly cheaper Surface 3 has a slightly smaller screen at 10.8-inches compared with the 12-inch of the Pro, is certainly lighter and has a slightly longer battery life, but also has reduced performance. Its Intel Atom 1.6GHz quad-core CPU and 2 or 4GB or RAM is likely to run perhaps two thirds as many plug-ins as an i5 Surface Pro 3, but if your audio projects are fairly modest then it could still be a candidate. It could also be a sensible choice as a portable multitrack recorder for live gigs when you don’t need to run real-time effects.

The Surface Pro 3 is currently available in five versions ranging from a 64GB storage Core i3 model which I would personally avoid for audio duties, through 128GB and 256GB versions with a Core i5 (rather more like it), to 256GB and 512GB with Core i7 — best of the bunch for running a sizeable number of plug-ins and softsynth voices, if you can afford the premium price. Their CPUs run, depending on the particular model, at between 1.5 and 1.9GHz. All Surface 3 models are now shipped with Windows 8.1 (the previous and ill-liked Windows RT is now defunct), so you can install AT 34

and run the vast majority of Windows desktop software without any restriction. You’ll also get a free upgrade to Windows 10 when it’s launched.

A DIFFERENT ANGLE

Microsoft’s Surface 3 and Surface Pro 3 only have a single USB 3 port, but adding an inexpensive hub (preferably a powered one) will give you more I/O flexibility, such as being able to plug in an audio interface, external hard drive for backup, iLok dongle, and so on. Microsoft does offer a Surface Pro 3 Docking Station with an additional three USB 3 and two USB 2 ports, along with various other I/O extras, although apart from the significant extra expense it does also seem a little perverse to end up transforming your Tablet into a desktop device in the process. The Docking Station’s fixed screen angle may not suit everyone either. Even the Surface 3 provides a selection of angles via its integrated three-position kickstand, while the Pro models feature an even more sophisticated multi-position kickstand that lets the musician set up the screen anywhere from near vertical to near horizontal, the latter perhaps being most suitable for using the touchscreen device to mimic a mixing desk.

No-one would expect any of the Surface 3 range to offer anywhere near the performance grunt of a modern desktop PC capable of 300 or more simultaneous plug-ins in a DAWbench test, but an i5 SP3 might manage perhaps 120 of them with a conservative buffer size of 256 samples, and maybe 70 or so from the Surface 3. When it come to audio performance it’s generally good news, but not without a few hiccups along the way. As with all mobile computing devices, there tends to be a trade-off between performance and temperature — there simply isn’t enough space in a skinny tablet case to install multiple cooling fans. Although beautifully engineered for burst performance, if the Surface decides that your audio software is pummelling its CPU too hard on a sustained basis and therefore getting too hot, it will significantly throttle down the processor speed until the machine cools down a bit, bringing your song to a stuttering halt. You’ll just have to be a bit more restrained with your plug-in usage to avoid such scenarios.

COMING INTO TOUCH

Of course the biggest news when using a Surface is its multi-touch capability. While built-in to Windows 8.1, it only makes sense to have it on offer when you have a touch screen, which is why so many musicians got frustrated with Windows 8 in the past. However, while Surface/Windows is a match made in heaven, when it comes to software multi-touch capabilities, it’s still not exactly a level playing field. Touch screens work beautifully with apps designed specifically for them, but when it comes to the DAW GUI, all bets are still off because most don’t specifically support touch-based screens. It’s a classic chicken and egg situation; if few musicians are using touch screens, why should a DAW developer go to all the time and expense of adding specific touch-based features to their software?

Thankfully, Windows 8.1 is written such that in most DAWs you can use single touches to select menu items, as well as swipe, scroll, pan and flick.

Many also accept two-finger pinch/zoom gestures, although drawing with the finger tip for creating automation curves and so on works in some applications but is ignored in others. Given the size of individual items on the Surface screen (buttons, sliders and so on), those with less than dainty fingers may still find the going frustratingly precise.

However, the Surface Pro 3 does include the beautifully-designed Surface pen (it’s an optional extra for the Surface 3), which makes selecting small on-screen items a lot easier. The pen also has a handy right-click button, so even the clumsiest among us may not have to resort to a mouse for much of the time.

Sadly, multi-touch audio software is still somewhat rare. Sonar X3 lets you mix with all your fingers, and even offers larger than normal effect panels for those with fatter digits, but most don’t.

A few third-party developers also offer adjustable size VST plug-in effects (such as Cytomic Audio’s most excellent The Glue compressor, and the entire Melda Production range), but again most don’t.

You could say we’re currently just a little touched!

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REVIEW

MOTU 1248 & ULTRALITE AVB

Networkable Audio Interfaces MOTU’s new AVB interfaces look a lot like the old ones. But inside there’s a whole new level of conversion and a blazingly fast network of audio.

NEED TO KNOW

Review: Mark Davie

PRICE 1248: $2949 Ultralite AVB: $1299 AVB Switch: $599 CONTACT Network Audio Services: 1300 306 670 or networkaudio.com.au

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PROS Super low system-wide latency.

Massive upgrade in conversion.

Up to 512 channels.

Wireless remote control via web app.

CONS Latency compensation between host and networked interfaces not automatic.

Lack of red clip indicators.

SUMMARY MOTU doesn’t change things if it doesn’t have to; case in point, its rack chassis. But inside, ESS Sabre DACs and new ADCs bring the conversion up plenty of notches over its older gear. And MOTU’s overachieving AVB spec lets you build a super fast network of interfaces without breaking the bank.

The standard AVB protocol specifies a fixed latency of 2ms, but MOTU has managed to claw back some digits for a fixed network latency of 0.625ms, which is staggeringly low

I’m a MOTU fan… for the dullest of reasons.

I’m a fan because, like its un-fashionable style — MOTU’s used the same black hard-wearing chassis on all its interfaces for over a decade now — its interfaces never go out of season.

In the same time as I’ve owned a MOTU 828 mkII, I’ve had to put numerous other interfaces — from all number of manufacturers — out to pasture. All the while, MOTU has silently kept updating its driver software to let me keep my 11-year old interface running. A friend of mine who owns an original blue-fronted, 14-year old 828 — the first ever Firewire audio interface (although Metric Halo may beg to differ with its 2882) — could keep it operating in the latest Mac OSX Yosemite or Windows 8.1 operating systems if he so chose. It’s almost unbelievable in this age of three-year product life cycles that the user is actually given a choice when they want to retire an interface, not have obsolescence foisted upon them.

Worth an ovation.

So here I am, looking at the new Thunderbolt and AVB-equipped 1248 interface from MOTU, in that all-too-familiar metal chassis, and wondering, what’s MOTU done different under the bonnet?

YOU’RE FIREWIRED

For one, the calling card of the 828, Firewire, is nowhere to be seen. While still available on the 828 mkIII Hybrid, it was dropped on the relatively recent Thunderbolt 828x. I’m fine with that. But I’ve also got another member of the family on review, the Ultralite AVB, and I know a few loyalists that will be disappointed at Firewire’s omission. The Ultralite has a reputation as a rock-solid playback interface for live shows, much to do with two

important features; it’s small for its I/O count, and it can be bus-powered via Firewire, eliminating the external power supply as a point of failure. MOTU has foregone this last point in favour of universality and I can’t hold that against it, Firewire’s long been on its way out. Even though Thunderbolt seems a more logical next step for a previously bus-powered device, I much prefer the Class Compliant USB 2.0 route MOTU has chosen — you can even plumb it into an iPad when needed.

1248, YOUR NUMBER’S UP

To be clear, the 1248 isn’t an 828 spinoff, it’s the lynchpin of MOTU’s new AVB family. It’s the ‘classic’ audio interface of the bunch: four mic pres, two DI inputs, eight line analogue inputs and outputs, 16-channel expansion via ADAT, wordclock and S/PDIF I/O, main and monitor outputs, two headphone outputs, and some meters and knobs on the front. Other than the AVB connection, you’ve seen it before. The other members of the clan are the aforementioned halfrack Ultralite AVB; the eight mic pre-equipped 8M; the 16A 16-channel analogue line AD/DA converter; the 24Ai 24-channel analogue line-in AD converter; the 24Ao 24-channel analogue line-out DA converter; the Monitor 8 six-channel headphone amp/monitor controller; and the digital-equipped 112D, which can plumb up to 112 channels of either 64 MADI, or 24 ADAT, SMUX or AES/EBU to and from your AVB network. The highlight, of course, is that they all speak to each other over AVB. Meaning you can easily upgrade your setup at any time with another interface that perfectly suits your new expansion requirements.

The family is comprehensive as it stands but

MOTU has shown no sign of slowing down the introduction of new siblings, so who knows what might pop out of the oven.

All the new AVB interfaces have a large blue LED screen that dominates half the unit’s width (in the case of the 1248). It’s a step up in versatility from the single LED metering of MOTU’s previous series — menus, larger readouts and full metering — but there are a few issues. For one, it’s a little hard to read at certain angles, so if it’s in a rack, getting an eyeful might require a bit of head movement. But more disappointing is it’s duo-tonal, white or black-over-blue scheme (depending on the model). Having only one colour to display information means you don’t get clear, easy indication of overs. To be honest, that’s about the only thing I ever use interface metering for. Just another red light out of the corner of my eye to let me know if something’s too hot. On the plus side, there’s a little gunsight icon in MOTU’s web app that blinks the display to let you know which unit you’re adjusting, or help you find it if it’s squirrelled away in a dark, closet server rack.

FAMILY PROTOCOL

The entire family has two protocols in common — AVB and USB 2.0. They’re divided into two main camps from there: interfaces with Thunderbolt (1 and 2-compatible), and those without. In the ‘In’ camp are the 1248, 8M, 16A and 112D. On the outer are the Ultralite AVB, Monitor 8, 24Ai and 24Ao. The thinking is that you’d always go for one of the former interfaces first, and they’re more likely to be closer to your computer than an expansion unit.

The exception being the portability of the Ultralite AVB, where USB 2.0 makes most sense.

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The upshot is that you can run 128 inputs and 128 outputs to and from your computer via Thunderbolt, up to 96k (half that at higher sample rates), and MOTU’s USB driver can run 64 I/O at 44.1k/48k, which is… frankly, incredible compared to most USB 2 interfaces.

There’s no restrictions on what you can route as an input. Oftentimes with audio interfaces, if you plug a mic in, you lose one of your analogue line inputs. It’s almost the opposite on the 1248, MOTU has so much bandwidth to spare, if you can plug it in, you can route and record it.

Routing is fairly easy to accomplish on MOTU’s web app. It’s a matrix setup with the ability to click and drag your cursor over multiple blocks to assign I/O in a hurry. The web app is locally served from the devices themselves. Once you download MOTU’s AVB manager, a little icon appears on your dock with a menu of any available devices.

Hitting a device opens the web app, where you can adjust routing, set device preferences, as well as adjust the gain of your preamps or trims of your line inputs and outputs, as well as control phase, phantom and pads. Every element is well set out, at first the gain dial seemed a little difficult to set accurately, but when you touch it a box pops up allowing you to type in an exact value; which can be up to 56dB on the Ultralite AVB, and 63dB on the 1248. There’s also a little gain meter above the dial, where the single colour scheme strikes again.

No red for overs. MOTU’s looking into that.

If you have just two devices, you can forego the AVB switch, and hook them up directly using a Cat5e or Cat6 ethernet cable. If you only have one device, you can use the ethernet port to connect to a standard Wi-Fi router for wireless control.

I had the luxury of reviewing the AVB switch as well, so I had both devices plumbed into it, with an ethernet output plugged straight into my Wi-Fi router. Over the last few years I’ve used this system as a ‘worst-case’ scenario for Wi-Fi ‘ease of use’. I’ve got an Ethernet over Power adaptor that works as a two-port router, mini Wi-Fi hotspot, and extends my internet connection. Most audiorelated devices, even though they’re only using the network for control purposes, do not play nicely with this arrangement. The MOTU gear had no issues whatsoever.

WIRELESS CONTROL

This remote control-ability is a central advantage of the whole AVB idea, or any networked studio gear. Being able to sit at the drums and adjust mic preamp gain from your iPad or phone is really handy. Of course, it depends on whether you’re using external preamps or not, but even then, you could still hook them up to your MOTU converter and see the input levels coming into that channel on your device.

What’s more, the analogue line trims are only supposed to give you gain of between +2 to +22dB. But in the web app, the trim on each channel actually ranges from -96dB to +22dB, and seems to respond as such. Of course, there’s still a maximum input level of +24dBu on the inputs. But it does give a degree of control I’m not used to seeing on line level inputs.

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The web app is clearly set out, and updates in a snap. There are four main tabs: Device — where you set clocking, control inputs and set your number of streams; Routing — a matrix connecting any input to any output; Mixing — a console for any input with insertable processing; and Aux Mixing — where you can send inputs to create ‘zero latency’ monitor mixes, or group channels to record.

For some reason, I was expecting a lag between changing, say, the number of AVB input streams on the Device page, and seeing it update in the Routing tab. But it’s instantaneous.

What’s more, it’s nice that the Routing matrix doesn’t display every possible input and output by default. You can enable and disable banks of I/O, the number of AVB streams (which are in banks of eight), the number of mixer inputs, and the number of to/from computer channels, to condense or expand the matrix at will.

ROUTING SAMPLE SET

The routing worked very simply. I’d route I/O over the AVB network from the Ultralite AVB into the 1248, and pump it into my computer

via Thunderbolt. The clocking locks in without a hassle, achieving sample accurate phase lock over the ethernet connection. As well as routing direct inputs to outputs, you can also send I/O in and out of MOTU’s internal DSP mixer. Which gives you HP filter, gate, four-band parametric EQ, and a compressor on each channel, as well as a global reverb, and EQ and limiting on the group output, master bus, and reverb return. There’s also three stereo groups, and seven stereo auxes you can use to mix headphone sends, submix tracks live, or use in a stage environment.

Pro Tools recognised the CoreAudio I/O… for the most part. I did run across one snag. When I was trying to send audio over the AVB network to the line outputs of the Ultralite AVB at the end of the chain, rather than coming up as ‘AVB Stream 1 1’, etc, Core Audio was labelling those outputs as ‘Out (Multiple Destinations)’. Resetting the 1248 to factory settings sorted that little niggle out. I’d put it down to the bad treatment I often give interfaces during a review — re-routing with sessions active, unplugging interface cables while still operational, hard quitting sessions, and turning the units off and on hundreds of times.

The MOTU devices performed pretty admirably despite all these ‘no-nos’.

DSP RETURNS

One thing worth noting is that there are limitations to what the DSP can handle. When the interface is set to 48k, you can run the full 48-channel internal mixer with every last effect, EQ and dynamic control instantiated. Turning up the dial to 96k restricts the channel count to 32, and severely limits the leftover DSP for effect processing, leaving enough power for just 13 full channel strips and no reverb. Once I added the reverb, I was down to a parsimonious two and a half channel strips.

There’s a meter for DSP usage, and once you get to the upper limits, a little dialogue box essentially tells you, ‘no more, sorry’ any time you try to add another module.

After nixing a bunch of processing to make way for the reverb, it was a little bit of a letdown. It had that ‘pebble bouncing over concrete’ digital grain.

The controls are flexible, and you can create your own presets with varying amounts of spread, predelay, reverb time, etc. But the underlying algorithm isn’t doing the effect any favours. I tried freeing up AT 39

even more DSP to see if that helped, but it remained the same. The EQ and dynamics were quite good though, and switching the compressor to RMS detection made it particularly useful as a quick level control without fussing too much. The limiter is an LA-2A emulation, and a handy bus processor.

You can create channel presets, and route mixer inputs from the channel, which is much easier than jumping to the Routing matrix. You can also drag across on/off buttons to instantiate a whole row of compressors, for instance. Another time-saving use of the click and drag motion.

LATENCY GAP

The latency specs on MOTU’S AVB network are extremely good. The standard AVB protocol specifies a fixed latency of 2ms, but MOTU has managed to claw back some digits for a fixed network latency of 0.625ms, which is staggeringly low. And that’s fixed over an entire network of up to 512 channels, with up to seven switches and hundreds of metres of cable. You have to add a skerrick of time for internal processing, but you can expect to shoot things around the AVB network in under 1ms when you factor in internal DSP, etc.

Measuring the actual roundtrip latency of the 1248 interface over Thunderbolt returned a result of 70 samples, which equates to about 1.4ms at 48k, plus double whatever buffer you’re using. It’s very good, and if you’re operating at 96k, with a low 32 sample buffer, your total round trip latency will be within that 1.4ms mark — super snappy. This puts it at level pegging with the RME Fireface 802 according to the latency tests Andrew Bencina recorded in Issue 105. The latency over USB is not far behind, and if you have a Mac running Yosemite you can also connect directly to the computer via an ethernet cable, but the performance is more computer dependent.

My main issue at the moment is the lack of delay compensation between your host interface (connected via Thunderbolt or USB) and your other interfaces feeding signal to it via AVB. It’s a common issue when expanding your I/O count via ADAT — there’s a discrepancy between when sounds arrive from your host interface and those connected via ADAT. When I split a signal to the analogue line inputs of the 1248 and Ultralite AVB, and recorded the result, the Ultralite AVB was returning the signal AT 40

25 samples later than the 1248, even though its signal was passing through the 1248.

I would have thought, given AVB provides more information about the device than a ‘dumb’ ADAT connection, MOTU could have delayed the host interface to time align it with other interfaces on the AVB network, but it doesn’t seem to be the case.

This is one area Avid has licked with its HD I/O interfaces, and Focusrite Rednet routes multiple interfaces to a computer via a switch to eliminate the host bias. 25 samples isn’t very much — about 0.5ms at 48k — and it’s a rare occasion this would create an issue; phase and alignment issues if you’re splitting stereo inputs over different interfaces, for instance. Something to keep in mind.

MOTU has implemented a workaround, which involves routing everything through the DSP mixer to align the latency. But it limits the typical implementation of the mixer as a Post-FX monitor mixer. Understandably, MOTU is responding to its user’s request for the lowest latency possible; an admirable quest. But I don’t see the conflict between letting a solo host interface operate at its fastest throughput, then matching latencies when an AVB expansion is connected.

RATTLING THE SABRE

The ESS Sabre 32 output DAC is the same top-ofthe-line chip used by the Apogee Symphony. The key point here is the 32-bit word length is lowering the THD+N and increasing dynamic range. In the case of the Symphony, Apogee quotes a dynamic range of 129dBA, which is the chip’s spec, while MOTU pegs its dynamic range as 123dBA, which is a more reliable measured spec. Noise is essentially undetectable. The other advantage is being able to use the digital main output control without losing resolution; a common criticism of 24-bit DACs, leading many to rely on external monitor controls. I would still probably tend that way with the 1248, just because I prefer having a big knob-type of passive controller in front of me.

But if the 1248 is in arm’s reach, it’s not a problem to use the Main Output pot, and pressing down on it mutes the output.

The DAC is really quiet. I couldn’t really detect any noticeable difference in the sound stage between either the Avid HD I/O or the MOTU 1248. There was no noticeable noise on the MOTU, which is an improvement over what I

remember of the 828 mkII. On the line input side, the Avid HD I/O outspecs the MOTU for dynamic range, 122dBA to 117dBA. But again, whether quoted or measured, I find these numbers somewhat negligible in normal recording — they’re both very good. I was evaluating a Shure mic kit against an Audix drum mic kit as part of my tests, and after going through all modes of evaluation, it only reinforced to me that the impact of using different equipment goes down exponentially the further you get down the chain. A change at the source was huge, likewise, different mics netted completely different results, but the general picture of the drums was still there. Then, using the MOTU preamps versus some Neves, the difference was noticeable, with the Neves smoothing out the response and adding some more body to the drums. Still the MOTU preamps performed really well and make great all-round preamps. Getting down to comparing converters, it’s extremely difficult to detect any differences when you get to this level. After comparing recorded results, mixes, and just general listening, I was completely satisfied with the reproduction of the 1248 conversion. The soundstage appeared the same as the Avid HD I/O, and likewise, I found no difference in the response and extension at both ends.

MOTU UPCONVERTS

While MOTU has often been a great buy for the price conscious, this time it has leapfrogged up a converter chip generation to the best of what’s currently available, while still keeping things affordable. I’m sure Black Lion will come out with a mod for those really intent on eking out the best from the units. Yet, with the specs already on offer, it’s hard to see the justification. On the hardware side, after running the 1248 all day the unit exhibited only the slightest change in temperature.

I’ve tested other interfaces packed with I/O that have almost melted into my desk. Even though I didn’t get to test the bigger units like the 16- or 24-channel converters, it bodes well. Combining the software and hardware upgrades with MOTU’s dedication to supporting its units over the extralong term, I’d have no hesitation jumping onto its AVB network and discovering a whole new, flexible way of recording.

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REVIEW

MACKIE PROFX V2

Compact Analogue Mixers Mackie’s analogue multipurpose tool: keep one in your car boot at all times.

Review: Christopher Holder

It started with a problem. I needed a small-format mixer to put next to a drum fill. I wanted our drummer to be able to plug in his in-ears or headphones, turn down the level of the wedge, and for it to be easy to revert back when he’s finished.

Which was when I noticed Mackie’s ProFX v2 range — I was especially attracted to the micro four-channel model. That was my conundrum, and no doubt you’ll have your own head-scratcher.

Yes, you may need an analogue front of house console for your pub, club, church… recording studio, even. And the new Version 2 of the ProFX range will fulfil that role more than competently.

But equally you may need something to address a less than conventional application.

SIX IS BETTER THAN FOUR

NEED TO KNOW

The ProFXv2 Series features three compact models — ProFX4v2, ProFX8v2 and ProFX12v2 — along with the ProFX16v2, ProFX22v2 and ProFX30v2, which offer four-bus architecture and dedicated inline channel compression on select channels along with the extra channel count. Each model packs a seven-band graphic EQ which can be assigned to main outs or monitor outs. Version 2

PRICE ProFX4v2: $299; ProFX8v2: $549; ProFX12v2: $649; ProFX16v2: $1099; ProFX22v2: $1399; ProFX30v2: $1999

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CONTACT CMI Music & Audio (03) 9315 2244 sales@cmi.com.au

PROS Whisper quiet preamps Useful FX Well priced USB input and recording.

CONS No ‘signal present’ channel LEDs, only overload Short-throw faders a bit dinky on the 22- and 30-channel models

SUMMARY Compact mixers are sold on features as much as they are on sound quality. The ProFXv2 excels in both.

introduces the four-channel and the 30-channel to the range for the first time.

There’s three-band channel EQ with HPF, a monitor send, global phantom power, individual mute buttons and pleasantly smooth 60mm faders (except for the ProFX4v2 which sticks with rotaries to conserve real estate).

It’s a feature-rich setup. There’s a good degree of I/O, including RCA in/out on one of the stereo channels, send/return inserts on the mic channels and a USB port around the back.

USB IN & OUT

The USB port provides for digital recording and playback. It means you can record a two-track desk mix directly to your Mac or PC, which is great for logging a presentation, as well as recording a gig or band rehearsal. Alternatively you can hit the space bar on your computer and use the USB as a digital input for streaming music directly to the house during performance breaks. In fact, Mackie provides a Break button for just such a purpose.

That’s right, it’s ‘Break’, as in ‘take a break’ between sets — hit the button, and it mutes everything except your USB input (which has its own level pot).

The USB1.1 recording is a fairly blunt instrument — it’s the modern equivalent of a two-track tap from the desk — but it’s useful nonetheless. Digital recording is a big reason for

many operators getting pressure to ‘go digital’ with their new mixer choice, while this ProFXv2 feature provides one more reason to stick with analogue, if that’s how you like to roll.

all the usual suspects present (reverbs, choruses and delays) and they’re all very serviceable. There’s a footswitch jack for muting the effects, which is another cool inclusion.

PREAMP REDESIGN

FIERCELY CONTESTED

The revamped ProFXv2 series features a new preamp design and it’s a doozy. The Vita preamp (based on the already excellent Onyx design) features a Class-A front end, “dual feedback stabilisation” and “bias current optimisation”, which all serve to keep noise to a minimum. (Mackie specs the eight-channel mixer’s Equivalent Input Noise (EIN) for a 150Ω source impedance, from mic in to insert send out, max gain: -125dBu.) During my test of the ProFX8v2 I compared four vocal handheld condensers all simultaneously plugged into the four preamps of the mixer, including my old friend the Rode S1, the classy Shure KSM9, the super-hot Audix VX5, and the classic AKG C535EB. The Vita preamp allowed each mic to demonstrate its tone and personality, with more than enough gain (50dB on offer). In fact, the lack of background noise was nearly my undoing – speaking through a powered wedge at what I guessed was a civilised volume, turned out to the coming of a 120 decibel apocalypse.

The other big version 2 drawcard is the update effects section. There are a choice of 16 presets with

The compact mixer market is pretty fiercely contested. Yamaha has some great options, Soundcraft are always strong in this area, Behringer of course, and others I’ve no doubt ignored or forgotten. Mackie’s ProFXv2 is there or there about with its pricing, packs some nifty features, and has some of the cleanest preamps on the market at any price.

Like I mentioned in my introduction, the ProFXv2 will solve your own particular curly problem. If you need a 16-channel analogue mixer for your pub to mix bands like it’s 1985… all strength to your arm. But just as likely, you’ll need a compact mixer for some other application unique to you and your circumstances. It’s likely to need to be thrown in the boot; used by others unfamiliar with the mixer; used in the absence of any outboard; used in totally unfamiliar rooms, plugged into combinations of mics, amps and monitors you may have never seen before. The ProFXv2 is likely to have the right frame size and the right mix of features to get you there.

— Christopher Holder

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AT 43 14/07/15 2:39 PM

REVIEW

SSL SIGMA

Analogue Summing Device What’s the plus side of having an analogue summing mixer? SSL reckons the answer lies partly in the digital realm.

Review: Greg Walker

NEED TO KNOW

Back in Issue 106 I reviewed the Phoenix Audio Nicerizer Junior and was impressed by what a simple, well conceived analogue mixdown path could do for a digital mix. Fast forward six months and another 2RU summing device has landed on my desktop, this time created by analogue mixing gurus Solid State Logic. The SSL Sigma is a very different beast to the Nicerizer however, coming with a feature list as long as your arm and the promise of becoming the nexus of your studio, via its digitally-controlled analogue systems and generous interfacing options. While I was pulling the Sigma out of its modest box I had to wonder how much further SSL can shrink down its large format console technology before the baby gets thrown out with the bathwater. I was

PRICE $5799 CONTACT Amber Technology: 1800 251 367 or sales@ambertech.com.au

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very curious to hear whether the Sigma could make my mixes more than the sum of their parts, and whether it could impart that oh-so-desirable SSL sound along the way.

OUT OF THE ETHER

There are a few basic concepts to get your head around with the Sigma. One is that it uses the first 16 channels inside your DAW session to control the analogue input gain levels going into the summing circuitry. It does this via an ethernet connection with your computer using either HUI, MCU or MIDI protocols. Moving your DAW faders you can control and automate analogue input gain while not letting go of your mouse or taking your eyes off the screen.

The second major concept at play here is that

PROS 32 channels of clean, highheadroom SSL summing Integrated DAW control of analogue input gain levels Extensive I/O and comprehensive master section Multiple device remote control via web browser Informative front panel display

CONS Web browser control reliant on network Software installation may cause heartburn Wall wart PSU Pricey

detailed control of all Sigma’s functions come via a series of tabs in its web browser interface.

Individual inputs are selected for mono or stereo channels, assigned to mix buses, panned, soloed or cut; to name but a few of the many functions that are controlled in this way. The Sigma can also be networked with most smart devices via a router, and future software revisions should allow full control from iPads and other surfaces. The front panel rotary encoder and user assignable push buttons allow a modicum of control over source monitoring and output gains but all the nitty gritty stuff can only be accessed via the web browser software and an internet connection.

To many of us versed in the old hands-on ways this may seem quite odd. You don’t need

SUMMARY An innovative software approach to hardware control, extensive I/O, and the monitor and master section features make the Sigma stand out from the summing pack. The Sigma wants to be the miniaturised analogue hub of your studio and mounts a persuasive case with great design smarts and superior sonics packed into a small footprint.

The Settings page is where you select your choice of DAW, assign MIDI ports, set up your IP network as well as selecting metering scales, SIP or AFL solo mode, and OK software updates.

The Master page takes care of metering assignment, mix bus insert selection, monitor and headphone routing as well as setups for Dim, MIDI learn and the user assignable button settings.

The Channel page takes care of mix routing, stereo/ mono switching, and solo, cut and panning for each individual channel, while a row of global commands down the bottom of the page allows for quick resetting of all 16 channels’ parameters.

to login to a website to adjust the routing scheme, panning, etc., on most analogue equipment.

Nevertheless SSL clearly sees advantages to this way of working — one big benefit being that the Sigma can be tucked away in a rack tower installation, where the operator can control all facets of the summing process without being physically close to the unit. Another benefit is in keeping the front panel very streamlined and informative using a variety of generous backlit LED displays. There is, of course, a large cost saved by not fitting the unit with large numbers of knobs and buttons. The asking price of the Sigma lies within the realms of possibility for the well-heeled home studio or small pro studio owner, while offering a good deal more than most straight-up analogue summing devices on the market.

ANALOGUE HEART

The Sigma offers 16 input channels selectable between mono and stereo as well as direct outs feeding two independent mix buses (Mix A and Mix B) that can also be fed into one another for mix bus parallel processing. There are mix bus inserts, external source inputs and a well-endowed

monitoring, headphone and talkback control centre. On the front panel are two customisable user buttons beneath a rotary encoder that selects between the various mix, monitor and headphone output levels. There’s also a mini jack plug for iPhone and laptop input and a 1/4-inch headphone socket. A ‘MIDI learn’ function allows further customising of operations via external MIDI devices and there’s a footswitch socket around the back that can be assigned to various functions but is nominally set up for talkback on/off switching.

It’s clear that SSL wants the Sigma to take over as the hub of your mixing workflow and be its beating analogue heart. In order to do this, a swarm of compact D-sub connectors on the rear panel covers the 32 channels of inputs, direct outs, mix bus inserts, talkback and even some miscellaneous sends and returns thrown in. In order to properly unleash the power of the Sigma and interface it with even a modicum of outboard gear, a comprehensive patch bay set up will definitely be required.

Two sets of monitor outputs and the Mix A output are on dedicated stereo XLR outputs, and there’s an ethernet port and a USB port for

factory diagnostics. The only thing letting down the general vibe of supreme compact 21st Century English Analogue Cool is the +12V DC wall wart power supply — unfortunate but true! SSL is quick to point out that the Sigma is built using the same SuperAnalogue circuitry that graces their Duality and AWS consoles and the popular small-format X-Desk. With a price tag well above $5k we’re talking a pretty big outlay here, so the Sigma needs to be judged with the spotlight on an eye for functionality and ease of use as well as a careful ear for sonic performance.

GETTING A GRIP

I’ve got to say my first day with the Sigma was one of the most frustrating I’ve had with a new piece of audio kit. Not that this was entirely SSL’s fault but nor was it all mine… bear with me. My studio is an internet-free zone and nestles in a mobile coverage black hole in regional Victoria. When I have bands here we have super productive days and we focus on music without any interruptions (there’s phone reception up the hill and internet in the house for those that really can’t hold on).

I also do a lot of TV soundtrack work and often have months of relentless weekly deadlines where technological issues and hold-ups are strictly off limits. When I really need to upgrade my software or register something new I put my iMac under my arm and go into the house, otherwise my work computer does not stray near anything resembling the internet. I rarely have computer hassles or new software-related bugs and so the days pass happily by. The Sigma was not conceived for people like me!

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Fortunately I keep my setup pretty portable as I often do location recording, and it was no big deal to make a temporary camp in a spare room in the house with Wi-Fi reception. Initially I assumed I would be quickly uploading some software and be off and summing within the hour. As it turned out there were quite a few curve balls to negotiate; including registering serial numbers, configuring internet protocol versions, updating the Sigma software while checking OS compatibility, as well as reconfiguring the HUI ports in my Pro Tools setup (turns out you need 40 of them, even though you will most likely only use two!); all with their requisite reboots and ‘handshakes’.

After about four hours of this I still wasn’t seeing full integration between my computer, Pro Tools and the Sigma and I kept coming back to the same dead end. After multiple read throughs of the user manual and a vain search for aid from various online forums, I finally discovered a small note in the manual about switching off Wi-Fi during reboot after loading the software from the website.

Of course the software that needs you to be online to operate the unit needs you to be offline to install it! Finally I was in and able to access the Sigma’s nether regions via the (by now much cursed) web browser but it was far from a smooth process.

LOCK & LOAD

Once you negotiate the twists and turns of configuring your Sigma software (to the supplied Pro Tools template in my case, but all major DAWs are catered for) the software/hardware browser interface is quite simple to use. There are separate pages labelled Master, Channels and Settings, as well as additional tabs for saving and loading settings into the hardware memory. The Master page takes care of metering assignment, mix bus insert selection, monitor and headphone routing as well as setups for Dim, MIDI learn and the user assignable button settings. The Channel page takes care of mix routing, stereo/mono switching, and solo, cut and panning for each individual channel, while a row of global commands down the bottom of the page allows for quick resetting of all 16 channels’ parameters. The Settings page is where you select your choice of DAW, assign MIDI ports, set up your IP network as well as selecting metering scales, SIP or AFL solo mode, and OK software updates.

The Pro Tools template gave me a basic setup which I then tweaked to my preferred way of working. I chose to output the audio from my converters in stereo pairs so I kept all the channels in stereo mode rather than switching them to mono and engaging the browser-controlled AT 46

panning. Hitting the Save button in the browser locked these settings into the hardware’s memory for ongoing use. Selecting Mix A as my go-to bus brought the master meters to life and I ran the Mix A output back into a stereo channel in my DAW. With the Sigma talking to Pro Tools, the first 16 channels of any mix will be highlighted to signal the implementation of HUI control. So it’s important to create 16 blank channels at the top of any mix to avoid double dipping when adjusting levels. Once this is done it is undeniably nifty to be be able to raise and lower the analogue input levels of an entire mix with one grouped set of faders or explore some interesting individual channel mix automation techniques. For instance, with an outboard compressor patched into your vocal channel you could automate the DAW’s output to hit the compression harder in a chorus and then manage the post-compression gain settings via the corresponding Sigma input channel.

Controlling output levels for the main mix, headphones and monitoring was very easy via the rotary encoder or Master web tab, while accessing features like mono fold-down and monitor switching through the user assignable buttons made for an easy, intuitive mix assessment process.

Full marks to SSL for the bright and informative front panel displays which keep things clear when navigating through the various options, as do the various controls in the web browser. Once up and running the Sigma feels like a classy piece of kit, providing all the requisite tools to be the control centre of your studio and taking up very little space while doing so.

GOLDEN HANDSHAKE?

So how’s it sound? You can be the judge yourself by going to the AT website and checking out the samples of a couple of different mixes I ran through the Sigma’s summing chain. For comparison I’ve included straight digital mixes as well as mixes summed through the Phoenix Audio Nicerizer Junior at low and high gain levels (at the latter setting that unit starts to introduce some noticeable transformer saturation and compression). There’s no mix bus compression on any of these samples and all the summed mixes were recorded back into Pro Tools using a Universal Audio Apollo interface.

The Sigma mixes do not exhibit overt harmonic saturation but that’s not really this unit’s game.

What it excels at is high headroom, wide soundstage sonics with excellent depth and clarity. For mine the SSL mixes sound the most refined and ‘expensive’ (which indeed they are). It’s certainly not chalk and cheese but the top end is particularly well defined and airy compared to the slightly more

AUDIO SAMPLES Listen & download at: tinyurl.com/ssl-sigma I chose a more open, acoustic jazz piece and a busy pop song to give some contrast.

The songs are Molly’s Jazz (featuring singer Maryanne Plenkovitch) and Doom Boogie Ticket by Machine Translations. Both tracks were recorded and and mixed at broadcast standard 24-bit/48k resolution and there are short gaps between each of the four source mixes. The order of mix samples is as follows: • Straight digital • Phoenix Nicerizer 0dB gain • Phoenix Nicerizer +8dB gain • SSL Sigma Go nuts and organise a double blind listening party on the weekend with all your mates!

grainy, midrange-orientated Nicerizer and the more clinical, in-your-face straight digital sound.

The SSL flavour is definitely there, not as emphatically as it would be through a 96-channel K series console bursting with analogue EQ and compression, but nevertheless it is present.

The subtle enrichment of the top end, the solid, cohesive low end and the revealing detail in things like reverb tails are hallmarks of that sound and it’s the reason SSL’s other more budget-orientated offerings such as the Matrix and X-Desk have such a solid fan base.

Workflow will be a decisive factor in determining whether the Sigma is right for your setup. If you like to get your hands on real faders and push a mix around then look elsewhere. If you don’t like dialling up web browsers to change panning setups in your summing mix ditto. If, however, you’re looking for outstanding sonics with a minimal analogue footprint and need 32 channels of pro-level summing with extensive I/O options to interface with your outboard collection, your number has just come up.

The Sigma won’t suit everyone but it shines at everything it sets out to do and significantly raises the stakes in the summing game with a powerful list of add-ons. The SSL sound is alive and kicking in the 21st century and more accessible than ever.

That’s something to be grateful for.

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REVIEW

RODE NTG4+

Shotgun Microphone It’s not ‘one number’ better than the NTG3, but with onboard filters, internal battery and a built-in pad, the NTG4+ has plenty going for it.

NEED TO KNOW

Review: Ben Osmo

Price $499

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Contact Rode Microphones: (02) 9648 5855 or info@ rodemic.com

Pros Internal rechargeable battery Onboard EQ & pad make it a flexible run-and-gun mic

Cons 4 is not better than 3

Summary The Rode NTG4+ has everything your camera doesn’t have in the sound department — highpass, high-shelf boost, and a 10dB pad — to let you capture sound in every situation. Its internal rechargeable battery also cuts the need for a phantom power source. But don’t be fooled; it’s not better than an NTG3, just different.

Location sound for film and TV is my specialty, and I’m always interested in loading up my blimp with a new shotgun mic, especially one with cool onboard features like Rode’s new NTG4+. It has three digitally-switched buttons with LED indicators. The top switch is a high frequency boost that starts around 3.5kHz, and the middle switch is a 75Hz high-pass filter to get rid of any boom rumble or handling noise. The bottom switch has a couple of functions. Firstly, it switches the mic on by providing phantom power from its internal rechargeable lithium-ion battery; the LED lights green when the internal battery is in use, and blue when using external phantom power.

Press again and it activates a 10dB pad for high volume sources. It’s always nice to have a pad inside the microphone, so the preamp doesn’t overload, particularly if you can’t pull the mic away or it’s attached to the camera.

The internal lithium-ion rechargeable battery is worth mentioning again. It’s very handy for devices that don’t have phantom power, such as entry-level mixers and most DSLR cameras. Which is where I think Rode has targeted this mic, but a bit more about that later.

SWITCH AROUND

The NTG4+ recharges in about two hours via USB, and Rode claims it will last 150 hours, though I didn’t test that myself. Once you plug into phantom power, the battery disconnects. But if you lose

phantom power or simply power down your rig, the battery will instantly power up the mic without a drop out. Just remember to power down the internal battery when wrapping your gear.

Outside my lock up, I set the mic in a Neumann suspension mount and put on the Rode wind sock.

I didn’t have too much trouble with handling noise, the high-pass filter worked well for that. When I switched back to flat, there was a light breeze and it still seemed fine without the filter. When the breeze came up a bit more, I switched the high-pass filter on again and the rumble was attenuated. Any more wind though, and you’d need at least a softie or a full wind shield and a wind furry cover. That’s where the high boost switch comes in handy, it compensates for the slight loss of high frequencies when covered with blimps and furries.

Once in a suspension mount, the NTG4+ can be mounted on a stand, a boom, a handheld pistol grip, or on a camera. I took it to a sound stage with boom operator Mark van Kool, and started an AB test. We were on a shoot, so there wasn’t time to do elaborate tests, but we set the Rode NTG4+ and one of my Shoeps CMIT5U microphones on boom poles and placed them next to each other. Mark talked into both mics and we paid attention to voice quality, timbre, frequency response, and back and side rejection. The NTG4+ seemed to have less bottom end, was a bit peakier in the mids and slightly harsher at the top end.

Obviously it’s not a fair comparison, as it’s not

in the same league as the Schoeps CMIT5U, and the price reflects this — it’s about a sixth of the Schoeps. But you can see my mistake. Looking at the model numbers, I assumed that being number 4, it <must> be better than the NTG2 and 3. But after inspecting the information provided by Rode, I realised the order was out of whack. The NTG3 RF-Bias condenser is a superior microphone to the NTG4 in its RF rejection, sound quality and ability to perform in all weather conditions.

The NTG4+ is actually positioned as an improvement over the NTG2 — a big one when you consider its EQ functions, the 10dB pad and internal rechargeable battery — but below the performance, sonic abilities and robustness of the NTG3.

The NTG4+ is a decent entry-level mic for run-and-gun shoots, and it’s particularly suited to camera-mount work, where the onboard controls and built-in phantom power let the mic do all its own heavy lifting. Obviously the NTG4+ doesn’t compare sonically with the high-end mics from Sennheiser, Neumann, Shoeps, and Rode’s other offerings, but it’s certainly practical and affordable.

The NTG4+ comes with an RM5 Mic clip, a foam windshield, a Micro USB cable and a ZP1 pouch. There’s also an NTG4 variant without the rechargeable battery, if you like the features but already have phantom power supplied.

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REGULARS

LAST WORD with

David Claringbold

Dave Claringbold has spent some 20 years working at the Sydney Opera House, most recently as Director of Theatre & Events. He has recently announced his resignation as he pursues other challenges.

In this second part of his recollections, David talks about the importance of people and culture to an institution like ‘The House’.

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When I returned to the Sydney Opera House as Technical Director in 2006, I shared senior management’s ambitions for the Opera House brand; ie. to be something contemporary, accessible and something that provided program for all types of people. Yet our own culture was holding us back from delivering that.

The Studio, when it opened, was the first venue we could truly call our own — we could invite artists and ask them what they would like to do. Trouble was, the culture at the time was almost militaristic. It was all about getting those lighting plots, rigging plots, and sound plots and delivering those plots to the letter. The new world was about an artist turning up, creating new work in the venue and our technical crew adapting and responding. It was going to require a huge cultural shift.

There were some quite fierce standoffs in that period. I needed the staff to know that I was in charge and I needed the customer to know that I was there for them. That we were there for them. The Sydney Opera House had just become so insular that it was all about our work conditions, our penalty rates, or whatever it was. All important parts of how a venue works but you don’t rub it in a client’s face.

It brought home the fact that technically we were a bit behind but culturally we were a long way behind. I knew then, that to get the funding to deliver the vision I had for The House, I would first need to address the human element of the equation.

One of the most significant changes was moving all the supervisors off a rostered, hourly wage regime, and putting them on a fixed annual salary. The message was clear: just deliver the gigs. There’s a lot of trust in that and it took them a while to get it: ‘we trust you and we want you to deliver the services as best you can’.

We also introduced new contractual arrangements that fitted in with the seasonal ebb and flow of the activity of the Opera House. We allowed casuals sick leave, recreational leave… I fought hard for those things. Without an understanding of what the crew needs in their lives how can you get anywhere? You can’t. Yes, I did go quite hard at management but I knew the staff on the floor were looking for leadership and I was there to give it to them.

The result was a better culture and better people.

After all, if we didn’t have the right people with the right attitude, with the right skills, in the right structure, any new technology would be redundant. If we didn’t have technology that aligned with the corporate strategy of the business, the technology would be redundant. Those things are inextricably linked.

To put technology investment on the table as a priority in an organisation, any organisation, you first must

understand the brand values of that organisation. Get hold of the strategic plan. Download the corporate or department business plan and use their words ‘against them’. “If our brand values are XYZ then our crap sound or crap lighting or crap staging is not living up to the brand values of the organisation, and we will not deliver on the strategic plan.” You need to look higher, from a strategic perspective, than ‘please may I have a new toy’.

At times I’ve even said, “no, we don’t need the money for that gear right now, please buy the new carpet”, for example. Showing you’re part of the team and that you can see what’s right for the business as a whole, at that time, will count in your favour.

Put together a three- to five-year plan. Show that you’ve thought about the future not just from a procurement perspective but from a maintenance, skills and training perspective.

Other times you just have to be plane sneaky.

I recall, around the turn of the millennium, that we inevitably needed to retire our TAC SR-9000 analogue consoles and move to digital. This was around the time Yamaha had launched the PM1D and Euphonix was launching the System 5. In anticipation, I knew we needed to run fibre throughout the venue. It would be an 18-month job. I snuck it past the bean counters by calling it ‘multicore’ in the budget — I didn’t actually mention the word ‘fibre’ for fear of the IT department having a meltdown. When questions were asked… “Oh yeah, that’s the multicore.” So it was kinda done by deception at first, which is funny given the way we now integrate with IT. From a network design philosophy, it’s like night and day.

The Sydney Opera House is now known as a place of technical excellence. A far cry from when I first darkened the doors in the early ’90s. To achieve that, you’ve got to start at the top. Everyone in the business has got to be living those values.

People who work in technical roles in the entertainment industries are all highly self-motivated, but you need to connect them into something bigger than “we’re going to do the gig right”. After all, you’re sometimes asking them to work extraordinary hours, hours that make no sense when you draft it out in preparation. You really want to plug into the motivation and the passion people have when they get into this business in the first place.

For me, it was about instilling the notion that “we’re part of something bigger”. That we were bringing new audiences and new artists to the Opera House, that we were changing perceptions, and how everyone in the technical teams were crucial in delivering that.

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