AudioTechnology App Issue 21 by Alchemedia Publishing

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Software

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Studio Microphones MACKIE MCU-PRO 9 Alps touch-sensitive faders, a full-sized backlit LCD and V-Pots for fast tweaking – the ultimate in hands-on command. AT 2

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REGULARS

ED SPACE Singing The Right Tune Column: Mark Davie

Editor Mark Davie mark@audiotechnology.com.au Publisher Philip Spencer philip@alchemedia.com.au Editorial Director Christopher Holder chris@audiotechnology.com.au Art Director Dominic Carey dominic@alchemedia.com.au Graphic Designer Daniel Howard daniel@alchemedia.com.au Advertising Philip Spencer philip@alchemedia.com.au Accounts Jaedd Asthana jaedd@alchemedia.com.au Subscriptions Miriam Mulcahy mim@alchemedia.com.au

I sounded laughably awful. And using my friend’s music room window as a mirror, I could see that I looked as silly as the noise I was making. My index fingers were shoved into my semi-inflated cheeks, and I was pursing my lips in my best approximation of a tuba player’s embouchure while attempting to buzz my way up and down a major scale. What came out could best be described as vocal flatulence. You see, my friend and I have embarked on what we’re calling a ‘lesson swap’. She — a professional singing teacher — is training me in the art of controlling my vocal cords. While I — impatient guitarist who failed miserably at trying to teach the only student I’ve ever had — try to give her a few pointers on how to handle six strings. In only a couple of these swapmeets, we’ve both already managed to have a few ‘a-ha’ moments. I’ve been singing (we’ll call it that, for the sake of this column) ever since I somehow landed a part as one of 12 brothers in our high school production of Joseph & The Technicolour Dream Coat. I really only got it because I was tall, and they were stacked for talent below five feet. I fudged my parts, and I’ve been doing it ever since. But I’m having to sing more these days, so I thought it was about time I got some professional help. The first thing we did was get a handle on my range, which delivered my first ‘a-ha’ moment. My whole life, I’ve technically been singing in the wrong octave. Not literally mimicking a castrato or anything like that. Rather, every time I’d played what I thought I was singing on a piano, I’d been playing an octave above what I was actually voicing. It was an odd moment, I must say, because I didn’t feel like I had any other major pitching issues — I naturally match the right piano octave on other instruments, and I typically reach for the right frequency, or thereabouts, when tuning on a graphic EQ. It just showed how unconscious the routine had become. At some point when I was younger, I’d picked up the bad habit of singing in the wrong octave and done it that way ever since. It’s exactly the sort of life-long mistake I’m trying to save my friend from. Distilling years of guitar playing that shaped my technique to save her having to go through the same teething process and potentially AT 4

develop bad habits. Unconscious movements to me — like how to direct your forearm so you’re applying pressure without hurting your wrist — feel completely alien to her. She would never instinctively go to the ‘right’ position if someone didn’t show her. But it’s already paying dividends; no sore wrists, better pressure application at her fingertips, no buzzing strings. There are often two ways we approach something we’re interested in. The risk takers go head first and figure everything out as they go. Others tend to research first or get someone to show them before attempting anything. Both can get you to the same place, but without a bit of solid guidance early on, you can end up singing the wrong tune. That’s what I like about our first tutorial from Ewan McDonald about how to use a measurement system to better understand your PA. He’s drawing on years of system tuning and PA development expertise, but he makes it clear that unless you get out there yourself with a PA and a measurement mic, it’s not going to make a lick of difference to you.

Proofreading Andrew Bencina Regular Contributors Martin Walker Paul Tingen Guy Harrison Greg Walker Greg Simmons Blair Joscelyne Mark Woods Chris Braun Robert Clark James Dampney Andrew Bencina Brent Heber Anthony Garvin Ewan McDonald Cover Image Corey Sleap Distribution by: Network Distribution Company. AudioTechnology magazine (ISSN 1440-2432) is published by Alchemedia Publishing Pty Ltd (ABN 34 074 431 628). Contact (Advertising, Subscriptions) T: +61 2 9986 1188 PO Box 6216, Frenchs Forest NSW 2086, Australia. Contact (Editorial) T: +61 3 5331 4949 PO Box 295, Ballarat VIC 3353, Australia. E: info@alchemedia.com.au W: www.audiotechnology.com.au

All material in this magazine is copyright © 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. 27/4/2015.

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COVER STORY

Chet Faker Producer Eric J Breaks Through with Built on Glass

ISSUE 21 CONTENTS

PA Tuning: Flat Lining your PA Could Kill your Mix

Studio View: Dave Clarke

RMS755

SUPER STEREO 10

GAIN REDUCTION

COMPRESSOR

THRESHOLD

RATIO

The Quick Mix: Tim Whitten

.15

2 .1 RELEASE-Sec

1.6 DUAL

GAIN

BYPASS

Yamaha DBR/DXR Powered Speakers

Audio-Technica M70x Headphones PC Audio

1.2

50 .05 ATTACK-mS

KRK Rokit 8 Studio Monitors

Apogee Ensemble Thunderbolt

Apple Notes

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GENERAL NEWS

AKG C314 MULTI-PATTERN MIC The AKG C314 is a new dual-diaphragm condenser microphone featuring super-low self-noise and dynamic range, a flat, high linear frequency response and neutral sound. According to AKG the C214 is the best selling recording mic in the US, so little wonder it wanted to fill the gap between the 214 and the legendary C414 (and its modern variants). Key features include: four selectable polar patterns; the same C414 XLS goldplated membrane for “high linearity and neutral sound”; overload LED Detection Display, which indicates ‘too

loud’ SPLs; and a usual array of accessories. Also of note is the integrated capsule suspension, reducing mechanical noise and resonances. A double-mesh, allmetal grille protects the capsule and ensures high RF immunity without affecting the microphone’s acoustical performance. A 20dB attenuation pad and bass-cut filter is included for close-up recording and reduction of the proximity effect. Australian Distributor Hills SVL www.hillssvl.com.au

SE RUPERT TUBE MIC sE Electronics has teamed up with Rupert Neve to produce the RNT large-diaphragm tube microphone. The mic features a three-position high-pass filter and ninestep polar pattern dial going from figure-eight to omni and everything in between. It has a high-voltage tube circuit within the microphone itself and Rupert Neve’s input is evident with the floor box containing a discrete, class-A amplifier circuit using the same custom op-amps as Rupert Neve Designs’ 5088 analogue mixing console. AT 8

There are also two of Rupert’s custom transformers and a custom mic capsule designed by sE to provide extreme headroom (153dB) and high-bandwidth frequency response. The ample headroom means there’s no need for a pad, but the amplifier circuit in the floor box has three gain settings of -12dB, 0dB (flat), and +12dB. The RNT will go for US$3495. Australian Distribution: www.sound-music.com

PRESONUS 192 TAKES COMMAND The Studio 192 is not just a USB 3.0 audio interface. Thanks to some nifty console ‘centre section’ features, Presonus likes to think of it as a Studio Command Centre. The new 26 x 32 interface records at up to 192k and combines eight digitally controlled XMax Class A solid-state mic preamps and Burr-Brown converters (for 118dB of dynamic range) with StudioLive Fat Channel signal processing. The command centre features are in its ability to manage speaker switching and talkback (with onboard condenser microphone), featuring main mix mute, mono, and dim. PreSonus’ remote controllable XMAX solid-state preamps use the same circuitry as the StudioLive AI consoles, but add a separate

digital volume-control circuit for digital recall without affecting the sound quality. The Studio 192 I/O includes: two front-panel mic/ instrument inputs and six rear-panel mic/line inputs; 16-channel ADAT optical in and out (eight channels at 88.1 or 96k); coaxial, stereo S/PDIF I/O; and BNC wordclock I/O. You get eight balanced TRS outputs, balanced stereo main outputs, and two headphone amplifiers with independent outputs and level controls. Australian Distribution: National Audio Systems 1800 441 440 or sales@nationalaudio.com.au

SOLID STATE LOGIC: SOMETHING IN THE AIR New LMC+ Module brings the classic SSL Listen Mic Compressor to the 500 format rack platform ... with added new creative features Solid State Logic has released the LMC+ Module for the 500 format modular rack platform. It’s a new and “significantly enhanced” version of the classic SSL Listen Mic Compressor. This famous processor from the legendary SL 4000E console was the accidental secret weapon in many producers sonic arsenal through the 1980s. Originally designed to prevent overloading the return feed from a talkback mic, its fixed attack and release curves were eminently suitable for use on ambient drum mics and delivers an aggressive and highly distinctive tone. While ambient drums are an obvious home for the LMC over time it has been used to process backing vocals, guitars, parallel compression feeds, and many other uses (and indeed abuses) have been found for its hard grab and fast bite.

wet/dry blend control, and the ‘Split’ button engages a bandpass subtraction mode that can compress certain frequencies of the signal and leave others untouched to be blended back together at the output stage. Using the ‘Scoop’ and ‘Split’ modes can subtly add interesting tone or completely destroy and deconstruct the audio you feed it. Australian Distribution: Amber Technology 1800 251 367 or professional@ambertech.com.au

This new version has a few new features: a pair of variable SSL high-pass and low-pass filters to allow targeting of a specific frequency range, a filter to compressor side chain option and a wet/dry blend control for instant parallel compression. Two additional new tools are added to give the LMC+ a distinctive sonic twist: the ‘Scoop’ button phase inverts the wet signal to give unusual sound sculpting possibilities using the

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LIVE NEWS

DIGICO S21 AFFORDABLE SHAKEUP Digico has released some big-time products over the course of its lifetime since the D5 changed digital live sound mixing, but you get the feeling that the UK manufacturer is about as excited as its ever been about a product release, with the new baby of the family, the S21. There’s no question the S21 will shake up the sub $10k digital mixer market, harnessing Digico’s use of the FPGA and its mature software to produce a powerful, affordable mixer. The list of hardware features include: 24 mic/line inputs, 12 analogue outputs, 96k operation, integrated USB2 audio I/O,

customisable bank and channel layout, two ethernet connections, 2 DMI expansion slots (up to 64 I/O per slot), wordclock, two AES I/O (mono). The software features include: 46 buses, user definable macros, a gate and compressor on each channel and bus, four assignable multiband compressors, four assignable DigiTubes, eight FX engines, 16 assignable 32-band graphic EQs, 16 flexi buses mono/stereo, 40 flexichannels mono/stereo. Australian Distributor: Group Technologies (03) 9354 9133 or www.gtaust.com

SOUNDCRAFT VI FOR ATTENTION Soundcraft’s Vi series has won plenty of friends over the years for their operator-friendly usability. There isn’t a person in the industry who doesn’t love the Vistonics UI. It marries controls and touchscreens better just about any other approach on the market. FaderGlow also pioneered the use of colour to more easily distinguish between output type, group, processing function etc. New to the Vi range is the Vi5000 and Vi7000 digital consoles, which replace the popular Vi4 and Vi6 consoles. The consoles partner a compact control surface with new Local Rack and Active Breakout box hardware. The Vi’s deliver simultaneous mixing of up to 128 inputs and 32 mono/stereo buses. Digital audio processing is at 96k, 40-bit floating point. Eight independent Lexicon multi-FX engines take care of effects. The Vi console also features BSS DPR901ii AT 10

integration and a BSS graphic EQ on every bus output. Monitor mixing has been streamlined and all buses can be turned to stereo without tying up two buses — great for IEM-intensive setups. Another cool feature on the Vi is the ability to monitor the status of any compatible Shure or AKG radio mic directly from the console surface (currently that is Shure’s ULX-D and QLX-D systems, and AKG’s DMS800 and WMS4500 systems). The engineer is provided with real-time visual displays of battery life, RF status, mic muting and internal clipping right from the Vistonics screen, avoiding the need for constant over-theshoulder looks to check the mic receivers — brilliant! Australian Distributor: www.jands.com.au

ADAMSON S: XL SOUND The S10 is Adamson’s two-way, full range line source cabinet equipped with two 10-inch Kevlar neodymium LF drivers and a NH4TA2 1.5-inch exit high frequency compression driver. It produces a slightly curved wavefront with a nominal dispersion pattern of 110° horizontal and is suited to a number of applications such as clubs, corporate events and churches. With reasonable speaker quantity, the S10 can handle theatres and arenas due to its increased vertical coverage of 10°. High frequency presence is maintained in the far-field thanks to the sound chamber’s efficiency,

and Controlled Summation Technology further eliminates low-mid lobing normally associated with twoway line source systems. The S119 subwoofer is the S10’s bassproducingcompanion loaded with a 19-inch Kevlar neodymium driver and a 5-inch voice coil. Both S10 and S119 cabinets are constructed with birch plywood and aluminium, fitted with Speakon NL8 connectors, and rigged using Slidelock technology. Australia Distributor: www.cmi.com.au

TF LOVE FROM YAMAHA We’ve seen the CL, QL and more recently the Rivage PM10 flagship, and now Yamaha closes the loop on its digital mixing range with the sub $6k TF series. The TF5, TF3, and TF1 respectively feature 33, 25, or 17 motorised faders along with 32, 24, or 16 rear panel analogue inputs featuring recallable Yamaha D-Pre preamps (for the first time in a digital console). There are a total of 48 inputs on the TF5 and TF3, with 40 on the TF1, including dual stereo analogue/USB digital inputs and dual returns. Live recording features include up to 34 x 34 channel recording and playback via USB2.0 and 2 x 2 with a USB storage device. TF stands for ‘TouchFlow’ which refers to Yamaha’s touchscreen-based UI. There’s also some nifty stuff like one-knob Comp and EQ control, there’s also an innovative GainFinder feature which has a smart crack at setting optimum gain for individual input signals — I think we all know someone who benefit from this feature!

Also provided is a range of input and output channel presets created in Shure, Sennheiser, and AudioTechnica, as well as respected engineers. The input channel presets are made to match a range of musical instruments and voices, covering parameters such as head amp (HA) gain, EQ, dynamics and more, right down to details like channel name and colour. Output channel presets include parameters optimised for in-ear monitors (IEM) and Yamaha powered speakers, with variations to match different environments and room sizes. All great stuff, especially for the novice. As you’d expect, TF allows you to do wireless mixing, personal monitor mixing, and offline setup with TF series consoles, there’s a bunch of SPX-sourced FX, and a expansion card slot. Pricing starts at A$3999 for the TF1 and $5999 for the TF5 and product will land in early May. Yamaha: au.yamaha.com AT 11

SOFTWARE NEWS

LAUNCHKEY LIGHTS UP Introducing the new and improved Launchkey MIDI controllers from Novation. Coming in 25-, 49- and 61-note sizes, the updated line now has RGB velocitysensitive pads that match users’ clip colours for more intuitive workflow. The Launchkey automatically integrates with Ableton Live to give instant hands-on control of your session view, instruments, effects and mixer. All three models have pitch and modulation wheels, buttons for transport control and eight rotary

knobs above the pads. The 49- and 61-note versions include nine faders for more comprehensive session integration. The units are USB bus-powered, saving you the hassle of lugging around a power supply. Launchkey comes bundled with Ableton Live Lite, Novation Bass Station and V-Station virtual instruments, and a 1GB collection of samples from Loopmasters. Australian Distribution: innovativemusic.com.au

CAKEWALK CATCHES RAPTURE Rapture Pro is Cakewalk’s all-new virtual instrument for Mac and Windows. Combining both algorithmic and sample-based synthesis, it comes with a 10GB+ sound library full of instruments, textures and soundscapes spanning a wide variety of genres. Based on Cakewalk’s Dimension Pro synth, Rapture Pro attempts to blur the line between oscillation synthesis and sampling so as to allow for vast sonic flexibility and tweakability. The user interface has a newly designed browser for easy previewing of presets created by top content designers

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from around the world. The UI also features a Perform mode, intended to connect the studio and stage with efficient controller mapping for real-time expression. Edit mode makes it easy to create signature sounds with extensive layering and mixing capabilities, and you have the ability to load your own samples. Rapture Pro will be available late April 2015. Australian Distributor: www.gibsonami.com

AKAI MIDIMIX MINI PRICE Akai Professional’s new MIDImix doubles as both a mixer and software controller. It has all the standard features of a MIDI controller with nine faders (one master fader), 24 knobs (three per channel), 16 buttons for mute/solo/record arm and one-to-one mapping to Ableton Live (Live Lite included). With the press of a button, you can send all parameters from the mixer

right into your DAW allowing you to mix and modify/ manipulate your projects concurrently in real time, giving you a much wider array of creative options as a result of the greater flexibility afforded by MIDImix. And the best part? RRP is under USD$100. Australian Distribution: www.elfa.com.au

IMAGINANDO’S FREE CONTROLLER Tech startup Imaginando have put out its latest creation – a tablet MIDI controller for iOS and Android called LK (Live Controller). Designed to integrate tightly with Ableton Live, LK is composed of four separate modules: Matrix (a dedicated, full-featured Ableton Live controller), MIDI Pads (16 MPC-style pads for playing

virtual instruments), MIDI Controller (general parameter control with four pages of 32 screen controls each) and X/Y Pad (a controller for LFO modulation). LK looks like a great tool electronic musicians will enjoy using both live and in the studio. Available for free download on the Apple Store and Google Play.

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REVIEW

Skinnerbox’s Time & Timbre Berlin duo Skinnerbox have built a harmonious two-part drum sequencer that’ll inspire you to break up that groove. Review: Jason Hearn

Skinnerbox is the marathon runner of Berlin’s electronic scene; improvising long live sets with fluid and evolving percussion parts. As it happens, the production duo, Olaf Hilgenfeld and Iftah Gabbai, are also developers who’ve released an innovative Max4Live instrument, Time & Timbre. It provides a fresh approach to creating drum tracks live and in the studio. It’s especially suited to Ableton’s Push controller, with an extended integration experience for sequencing and tweaking your beats! Before getting started you’ll need to have at least Live 9.1.6 installed and update to Max4Live 6.1.9. Owners of Push should check the controller is selected in the very topmost slot of the MIDI configuration page. Time & Timbre presents itself as a suite of Max4Live Instrument devices. Within Live’s browser you’ll find Skinnerbox Time (an XoXstyle drum sequencer) and five drum-dedicated synths prefixed with ‘Timbre’: BassDrum, Cymbal, HiHat, Snare and Tom. The Timbre instruments are serially inserted in a drum rack after the Time instrument, so you can modulate them with Time’s LFOs. Sound like fun? Indeed it is! HAMMER TIME!

Time is a six-track step-based sequencer. Where it departs from the norm is in its pattern randomisation functions and independent time/ roll metrics for each rhythm track — polymetric mania, ahoy! As you change a part’s meter, the steps graphically shrink or grow to maintain alignment relative to the other tracks. And each track can be AT 14

independently cleared, randomised, shuffled left or right, and play forward or backward. The global Roll setting allows not only different divisors but also different styles such as Gate, Decay, Reverse 1, Reverse 2 and Random. Finally there is a Global swing function (which can link to the groove of other Time instances), a Master Random function and a ‘Morph Master Mike’ parameter, which allows crossfading between A and B states of all controls. The four LFOs in Time can be mapped to Timbre’s synthesis parameters and any tweakable parameter within your project (perhaps to the cutoff frequency of Ableton’s Auto Filter inserted over the resulting drum bus). Each LFO sports a variety of wave shapes and can be independently set to sync to the project’s tempo or be free running with a maximum rate of 20Hz. TIMBRE CUTS

Timbre’s character is unashamedly synthetic in nature and would readily appeal to producers in genres such as Moobatron, Minimal House/ Techno, Trap and derivatives of Future Bass. For those familiar with the synth drum modules added in Native Instruments’ Maschine 2.x update, the Timbre modules are of the same quality however perhaps not quite as versatile in their overall character, or venture into physical modelling. Having said that, there are oodles of synthesis parameters on offer, some venturing into more experimental territory, including a parameter randomisation function on each module. In fact, the developers encourage using the random

function to find sounds in preference to browsing. With a few clicks I could go from abrasive ’80s game console chip-sound percussion to cleansounding vintage synth drums. BEATS IN PIECES

By splitting the two components, Skinnerbox allows you to use either module independently. The Timbre modules can be hosted outside of the Time sequencer, allowing you to play chromatically tuned bass line parts with the KickDrum or Tom from within Live’s MIDI clips. Future Bass producers — look out! Likewise, you can use any sampled drum tones hosted in Simpler/Sampler with the Time sequencer, or even 3rd party plug-in instruments such as XLN Audio’s Addictive Drums if you’re after more of a rock/acoustic edge. You could even pipe the MIDI data into the hardware realm of your studio with a vintage drum tone module. You could even go as far as building a melodic step sequencer with synths in the drum rack slots and create some Complextro acrobatic contortions, with the bonus of exotic grooves, swing and rolls being generated in real-time. The real essence of this product is how it handles as a live performance tool for rapidly putting together beats on the fly. If you have a thirst for a fresh approach to making beats that play to aesthetics of today’s electronic music genres, drink from the Skinnerbox cup. Get the pack for $55 from ableton.com

MAKE

GREAT MUSIC KRK Systems is one of the world’s most respected manufacturers of Studio Monitors. In our California state of the art design facility, KRK engineers create products that deliver a natural and balanced spectral response with low distortion, tight bass and superior imaging. With KRK monitors, subwoofers and headphones, recording engineers and artists hear every nuance of the audio being reproduced regardless of musical style, genre, or particular mixing needs. Please take a closer look at KRK´s line, used by many Gold, Platinum or Grammy award winning producers for tooling and crafting their great music.

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Focusrite Rednet

Neve 4081

Neumann U47fet (SPECIAL) AKG C414 AudioTechnica 5040 Sennheiser Mk4 Rode NTR SE Rupert Neve RNR1 500 Series

Neumann KH310A

Genelec 8250A

SSL Xrack API 3124 Dynaudio BM6 MKII UA 4710D Focusrite ISA 828

SSL Bus Comp (SPECIAL) API, Chandler, Neve & Focusrite Red

1 Nati o nal Park St Newcast le West N SW 2 3 0 3

02 4929 2829

Yamaha MSP7

w w w. m us os c or n e r. c o m. a u

90 Degree Studio Academy Presents MAY 2015 STUDIO MASTERCLASS Designed For Professional Sound Engineers, Producers and Students For the first time, a world-class master engineer and producer invites you to share a world class studio with them in this unique series of studio sessions that lets you get inside their mind and their mix. With Rob Taylor (Over 25 years experience, Chief Engineer Alberts Studio and The Grove, 3 x nominee for ARIA engineer and producer of the year, Current PhD candidate in Communication and Media Arts) The Masterclass Series is four courses - The Mix Journey (6 sessions) - From Mix to Master (2 sessions) - Masterclass on EQ (1 session) - Masterclass on Compression (1 session) Go to www.90deg.com.au/masterclass to see the full course listings and more about Rob Taylor. Enrolments are strictly limited. Criteria for entry applies. Cost : $110 per session (Less for full course enrolment)

Musos Corner & 90 Degree Studio 1 National Park St Newcastle West NSW 2302 02 4929 2829

www.90deg.com.au | www.musoscorner.com.au AT 16

REVIEW

AUDIO-TECHNICA M70x Headphones

The detail of the new M Series flagship won’t spare your feelings but may save your mix. Review: Christopher Holder

It’s easy to see why Audio-Technica’s M50s have been so popular. They manage to strike the perfect balance of price and likeable sound quality. ‘Likeable’ because the M50s are accurate enough for day-to-day studio use but exciting enough to make any well-produced audio recording enjoyable to listen to. For most of us the $200 M50s will be an ideal choice. But we may look longingly at pricier headphones and wonder what we’re missing out on. Well, don’t get too ‘grass is greener’, because pricier cans may be more accurate but not always as fun to listen to. I’ve had a pair of open back Shure SRH1440s for a couple of years now. They’re an amazing headphone — natural, accurate and beautifully detailed. But if I’m listening back to one of my EDM mixes they’re a real party pooper. The ruler flat response does nothing for my low end. In fact, it’s sometimes really quite hard to hear the full character of synth bass sounds. I’m much better off with an M50-style headphone. Saying that, the more detailed 1440s give me some unnervingly good insights into just about every other aspect of the mix. Audio-Technica has released its ATH-M70x which are the new flagship headphones of the M Series. They’re a $400 headphone, and although Audio-Technica avoid using the ‘reference’ word, everything that’s coming out of Japan reinforces the fact that these headphones are built for accurate, linear monitoring (all the way from a quoted talk-to-the-sperm-whales 4Hz to terrierterrorising 40kHz). Plug them in and you instantly get the picture: totally un-hyped low end, super-detailed mids and highs, and an extraordinarily detailed sound stage. Plug the M50s back in and, by comparison, they’re like a party going on in your head. Mixing with the M70x headphones is a sobering experience. You’re not spared anything. There’s no low-end, feel-good

distractions, it’s all detail: that reverb tail isn’t doing it for me; why am I hearing that vocalist’s bracelet jangling?; I thought I nailed that delay time?; I really wish I could get more width on that acoustic guitar; that double tracking could have been better… Want me to go on? But this is the intent of sticking headphones on in a control room. Detail. And the M70x delivers. Given the M70x is a $400 headphone, you’re unlikely to leave them sitting on a chair in the live room. But being closed back, you can record with them without fear of spill into open mics. In fact, for the self-contained producer this is a real advantage. The M70x may well be all the headphone you need — tracking, mixing, mastering… okay, maybe invest in a pair of hyped DJ headphones for when you’re on the bus listening to your iPod. (After all, let’s not deny ourselves the occasional ‘ignorance is bliss’ descent into overblown, outta whack performance.) The M70x isn’t for everyone. For professionals accustomed to open-back reference headphones, they’ll acquit themselves well. Up against my open back SRH1440s they felt largely cut from the same cloth. If this was an audiophile magazine I guess I could nit pick, but the more I listened, the more I realised that for a closed back headphone, the M70xs had far more in common with open backs. And here’s where the M70x has found its own niche. If you need reference-style headphones with open back attributes, but you need closed back privacy, then the M70s are a walk-up winner. They’re priced well, built well and look professional. Well worth an audition, but hold onto those M50s — life’s too short to be totally lost in the detail. Price: $399 Technical Audio Group: (02) 9519 0900 or info@tag.com.au AT 17

FEATURE

Chet Faker’s Built on Glass just keeps on piling up the accolades — ARIA Album of the Year and topping Triple J’s Hottest 100. For Eric J Dubowsky, it’s a mix that’s cemented the American amongst Australia’s elite. Story: Mark Davie Photos: Corey Sleap

Artist: Chet Faker Album: Built on Glass AT 18

Awards might not mean much to some people, but to Eric J, winning ARIA Engineer of the Year was validation for five years hard work that began with cold calling in a foreign country, and included a second act mix-off against Tony Maserati that won him the favour of the cutting edge Future Classic label. Five years ago, Eric was new to the country; an American with an Australian wife, who’d moved over to have their first child. In The States, freelancing in the music business doesn’t come with health insurance benefits, and the first round of tests and ultrasounds were already mounting up into the thousands. They needed to get out of there before they were loaded with debt, and Medicare was looking pretty darn good from across the Pacific. In his past life, Eric had engineered some notable sessions, including Weezer’s Red Album under the nose of Rick Rubin. But his American credits didn’t necessarily translate to the level of work he was accustomed to. He still had to prove himself to the cadre of Aussie music industry A&R execs. So with a diary of free dates, he sat on his new couch in Sydney watching the Australian record industry’s night of nights and started doing some homework. Each acceptance speech read like a PR hit list of prospective clientele. Direct from the horse’s mouth he was getting a verbal description of the owners, trainers and jockeys surrounding each stud farm corral. As each artist got up to receive their award he was frantically scribbling each set of names down. That night he sent the list to his manager in the States with a memo to put the call out. ALL ABOUT THAT BASE

It was an inauspicious start. For a couple of years he engineered, produced and mixed a range of modest projects, with all his Stateside professionalism intact. He was working out of a cubbyhole at Studios 301 in Sydney and started to break through, including a big stint working as a producer for The Voice. His big break onto the Australian electronic music scene came when he was recommended to record Emma Louise’s vocal for the Flight Facilities’ song Two Bodies. After recording in Studios 301’s main live room, the next day he took the Flight Facilities team upstairs to his studio for the edit. He gathered the whole crew upstairs, because he figured most laptop producers were working at home in smaller studios, so he wanted to show what he could do with access to outboard gear and a good listening environment. Plus, he said, “I know things about audio from years of just doing it. I was excited to get in with these guys because I feel like there’s so much exciting music coming out of Australia. I really wanted to work on some of this electronic stuff, but they usually don’t give you a chance unless you have electronic credits.” Eric’s account was low on those kinds of credits. But the show ’n’ tell paid off, Flight Facilities were impressed enough to give him the mix too. It became the single about eight months later. From there, Eric started checking out more electronic acts, and the trail kept leading back

to Future Classic. Luckily, he had an inside man. Jimmy from Flight Facilities had no qualms recommending Eric to Nathan from Future Classic. And from there… it snowballed. Eric: “I got a text from Jimmy saying, ‘Nathan from Future Classic is going to call you.’ Ten seconds after I got that text, Nathan called and told me, ‘I need a Flume mix done, can you do it today?’” He immediately cancelled his day, got the files sent over email, and started mixing the Flume track The Greatest View, which featured vocals from Isabella Manfredi of The Preatures. With the mix in the bag, Eric was asked to be the Musical Director for Flume’s ARIA performance of the same song. MASERATI TEST TRACK

By that time, he’d endeared himself to Future Classic enough to get a shot at an album, Chet Faker’s Built on Glass. There was one caveat: ‘Would you be willing to do a spec mix? Because the American label wants Tony Maserati to mix it.’ “Obviously I know that guy is badass,” said Eric. “But I was definitely up for the challenge, so I said, ‘Absolutely, give me the files’.” The test track was Talk is Cheap. Eric took a listen to the album roughs and realised it was, “a really great album. They don’t come along every single day, so I put everything into doing that first mix.” Like most spec mix-offs, he didn’t have any communication with Nick Murphy (who is Chet Faker). “It’s really stressful when you do a spec mix like that with no feedback,” said Eric. “No one to ask how much reverb they like or if they have any references.” The only place Eric could go for insight was online, where he dug up previous Chet Faker releases to get a sense of how much reverb to use. “Because that’s usually the big thing with singers,” he said. “How much ’verb they like on their voice.” He did his due diligence, plotting a course from Chet Faker’s previous EP and predicting where he imagined the artist would want to go sonically. “I knew it couldn’t be too polished because that’s not the vibe. But I still wanted it to be polished enough so it would sound really warm. As a mixer it’s your job to help realise that person’s vision, to take what they’ve given you and make it better. Especially since I KNEW NICK PRODUCES THE STUFF, SO IT’S NOT LIKE I’M GOING TO GO 180 DEGREES IN A DIFFERENT DIRECTION TO THE ROUGH MIXES. I made a choice to hold everything

out in one section towards the end and I was thinking, ‘Oh man, I hope this isn’t too much.’ I was really happy when I got the call saying everyone really liked it.”

BOXING DAY SPECIAL

He beat out Tony Maserati and got the gig. But it had taken him four days to nail that mix, and Future Classic wanted the rest of the record mixed in 10 days! Eric: “I started the next day, on Boxing Day; I was supposed to have the whole week off. I told my wife I thought it was going to be a really big record, and that I had to do it. I had to get somebody to turn on the circuit breaker at 301, because the studios were supposed to be closed! “I mix a lot of records remotely these days, sometimes I never meet the artist, we might just

Some elements you feel more than you hear, and that’s a big part of mixing to me

”

email a couple of times and that’s it. But it was down to the wire, and Nick was in Melbourne, so I asked if he could come up for five days just to save sending files back and forth. “Sometimes I like to have the artist there to ask specific questions. With Nick it was really imperative just so we could get it done quickly. When people bounce stuff out of Ableton Live, you don’t know if the files are exactly the way they’re supposed to be delivered — there might be a bell sound missing, for instance, an automation issue, or a wet/dry balance just not coming through properly. “With 1998, we couldn’t get Live to bounce the parts the way they were in his rough mix. It was literally down to about 11 o’clock at night the day before we were supposed to master it, and he was still at his laptop in the studio trying to get these files properly bounced down.” ATLANTIC SOUND

Eric has toyed with the idea of mixing in Ableton Live, considering most of the projects he’s mixing for Future Classic were created in the DAW. But he’s pretty happy with the sound of his Pro Tools HDX rig, clocked with the Isochrone 10M and Isochrone Trinity atomic combination. He has a bunch of outboard gear including his much-loved Bricasti M7 reverb, a Lisson Grove R-124 compressor, Moog analogue delays, Neve and API preamps, and AT 19

Top: The drum routing on the mix for Talk is Cheap shows how far Eric takes the Ableton Live laptop production. Left: The Antelope Audio Isochrone 10M and Isochrone Trinity atomic clock combination keeps Eric’s digital world in perfect time. Right: In his 500 series rack — alongside the Neve pre and EQ, and API 512c preamps — are two Moog analogue delays that get a real workout to create excitement in any mix.

listens to Barefoot MicroMain 27s alongside his Avantone Mixcube. One big piece of gear that Eric plans on integrating, so he can mix in other DAWs, is the mint vintage Harrison 32/32 console sitting front and centre at his new Hercules St Studios digs. “People say the biggest problem with Ableton is the panning isn’t as wide and the stereo field isn’t quite as three-dimensional,” said Eric. “The reason a guy like Rich Costey can mix in the artist’s preferred DAW is because he’s going out to a console. So the stereo summing doesn’t get affected.” Eric worked with Costey on the Weezer record, and what he took from that time was an appreciation for Costey’s irreverence; a complete lack of concern for ‘the rules’. “The lack of a system is what I found inspiring,” recalled Eric. “He didn’t do the same thing every time.” His earliest mentor though was Arif Mardin, one of the great Atlantic Records producers, and responsible for the ‘Atlantic sound’ as a key part of a triumvirate that included Tom Dowd and Jerry Wexler. Eric started working for Mardin at 23, AT 20

when the producer was in his late 60s. But he still had plenty of sage advice to go round. “It’s young kids that make electronic music, but historically it’s the guys with experience who work with those people,” said Eric. “Arif used to tell me he’d go to Studio 54 and just watch what the people are enjoying. Which is one reason why I’m going out to see Flight Facilities tonight. You have to be aware. There’s an equation; there’s the music, but there’s also the audience. “Then working on a Rick Rubin project, there were certain aesthetic principles that guy adheres to, and consequently, all the engineers adhere to. There’s got to be a human element and it’s got to affect you emotionally — it’s the most important thing. There are core principles I’ve taken from each of those people, but you kind of assimilate it into your own. “With Chet Faker’s music I’m trying to create more of an emotional connection to it. Sometimes laptop producers might be recording in a less than ideal environment. It can give it a vibe, but then my job is to try and eliminate the veil between the

music and the listener. So a big part of what I do is retaining the vibe, but getting rid of noise, ground hum, and clicks and pops that take you out of that other world. “THERE’S GOOD AND BAD NOISE, IT’S ABOUT KNOWING THE DIFFERENCE; FIGURING OUT WHERE THE VIBE IS COMING FROM AND WHAT IS PUTTING A BARRIER UP. It’s not the most glamorous of jobs, I spend quite a bit of time going through the tracks making sure there’s nothing that takes my head out of the emotion. “It’s like what Rick Rubin would say, ‘if you put a microphone in front of Johnny Cash, all of a sudden, that’s genius.’ Really, what he was saying was when you have a great artist, you just have to make it pure and stay out of the way. With Chet Faker, it’s obvious Nick worked really hard before I ever met him, and the songs are effortless when you let them become what they should be.”

KICK BACK & CHILL

The vision for Built on Glass was the kind of old soul that perfectly complements late afternoon summer sunshine and a G&T. So for Eric, the foundation

was always going to be the low end. “I’m from the New York area,” said Eric. “And the first New York City studio I worked in was a hip-hop studio. So the low end is the foundation everything is built on, especially with electronic music.” Even though the songs were leaning towards the mellow end of soul, it’s still “all about having that really focused kick, sitting right at the edge of the speakers,” he continued. “People have a different aesthetic depending on where they come from and what they grow up with. The classic New York sound was to always have the kick and bass really loud. ON ALL THOSE OLD ATLANTIC RECORDS

the transients would get rounded off by the tape, I wanted the kicks to sound really analogue and warm. I do a lot of parallel compression, especially on the kick, and I’ll have multiple compressors just for the kick. “I’ll bus all the drums to two parallel compressors, one with a really heavy compressor that I’ll only mix in a little of. Then I have a kick sub that I’ll move up or down depending on the part, the song or the genre. “There’s so many elements of the kick. YOU DON’T

THEY USED A VU TO MEASURE HOW LOUD THE KICK WAS

OF HARMONICS SO EVEN ON LOWER LEVEL SYSTEMS YOU

AND PUT THE BASS ABOUT HALF A DB LOUDER THAN THE

CAN HEAR THE KICK.

KICK. IT WAS A SCIENCE.

“I have a few different auxes with different compressor flavours to mix and match between. The plug-ins I use a lot on those auxes are the McDSP Compressor Bank, the Softube Tube-Tech, and the Valley People Dynamite — some might be for a little more bite, others for sub. And one of my secret weapons has been the Overstayer Saturator which is an outboard piece of gear. It’s like a

“I spent about half a day going through the sounds and samples on Talk is Cheap making sure the kick was in the right place. It hits you, but it’s not too punchy. “The kicks couldn’t have too much point on them because we were thinking about it like an old soul record. Knowing how, on those old records,

WANT THE SUB FREQUENCIES TO OVERWHELM EVERYTHING AND EAT UP ALL THE HEADROOM, SO I TRY TO CREATE A LOT

Thermionic Culture Vulture or a Decapitator — it really crunches things up in a nice way, and gives it some harmonic distortion. “I also add kick samples. Like everybody, I have a huge sample library, which is one thing that’s different about production now. When I first started it was ‘a thing’ if you had a good sample library. You used to seriously guard your drive that had samples on it, and wouldn’t let anybody copy your samples. Now in five minutes you can go online and get all of J Dilla’s drums or all of Jay-Z’s samples. It’s not a badge of honour like it used to be.” TRANSLATING VISION TO SOUND

There’s a second emphasis in conveying the artist’s vision that takes just as much experience. Eric: “I’ve heard songs I’ve mixed played back on lots of different systems — I’ve heard it on the radio, I’ve heard it in gas stations — so I know what gets lost in a mix. That’s another part of my job, to make sure the vision translates across all mediums. People are watching videos on YouTube and listening on their iPhones. I got this plug-in so I AT 21

A big part of what I do is retaining the vibe, but getting rid of noise, ground hum, and clicks and pops that take you out of that other world

”

could stream to my iPhone and any laptop. I have the little Avantone speaker, really nice Barefoot 27s which sound amazing, but then I also have a little Bose sound dock. I have every kind of speaker you can have.” And, of course, there’s also the car stereo. Eric was checking out one of the masters and realised the snaps were too loud on one song. “I’d checked on every system and then I heard that, ‘Dammit!’” he said. “I was half way home, but I drove back to the studio at two in the morning to fix it because it would have driven me crazy.”

IN TUNE WITH THE INTENT

One of the striking things about the lead vocals on Built on Glass is the lack of any tuning. It’s not AT 22

because Eric isn’t capable of working the controls on Melodyne, on any other record those warbles would be taken right out. But for Chet Faker it’s the sound of authenticity, and that vision of “a classic soul record. The record just happened to have some samples and some electronic elements on it, but at its core it’s about a guy singing and playing,” said Eric. “If the performance is authentic then people don’t really care. A lot of the kids today don’t necessarily know what the influences might have been, what old soul music might have sounded like. So it’s a point of a difference that it’s not overdone and lets the emotion come through. A lot of our favourite records are far from perfect. “It’s so easy to get this hyper focus. I’ll do a critical listen where I just listen to the vocalist singing the song to make sure nothing is taking away from that. AND ONE OF THE LAST THINGS I DO IS WHILE I’M LISTENING TO THE MIX IN THE BACKGROUND. MY

and depth to arrangements without introducing a new instrument to draw away the focus. Eric loves this arrangement style, treating backing vocals to an elaborate diet designed to fatten them up and widen them in the stereo field. It’s a creative process helmed by his Bricasti M7 reverb, analogue spring reverb and Moog 500 series analogue delays. He’ll often print loads of effects from the short and simple natural room reverb, to the wild and manually manipulated, working them all together into a subtle atmosphere. Eric: “BVs are now as important as they’ve ever been. It’s one of my favourite things, to use the whole stereo field and a lot of delays to place things around the listener. I’ve been treating the reverbs by distorting them, sending them into delays and compressing them. There’s so many things you can do, like side chaining reverbs. I try to make it so the vocals still sound upfront, but then there’s atmosphere around it. THAT WAS ANOTHER TECHNIQUE

MIND DISENGAGES, SO IF ANYTHING STANDS OUT, THEN

ARIF MARDIN SHARED WITH ME; SEND SOMETHING INTO A

THAT’S SOMETHING I KNOW NEEDS TO BE FIXED.”

HUGE CRAZY REVERB AND JUST BARELY HAVE IT THERE IN

READ AN ARTICLE IN THE NEW YORK TIMES OR SOMETHING

BACKGROUND TEXTURE

Depth is a big part of any mix, but especially in neo-soul, where there’s often a foundation of drums and bass, vocals over the top, and a lot of space in between. On this record especially, said Eric, “it was like kick, snare, vocal. As long as all that was in the right place, then you could have fun with the atmosphere.” Backing vocals can often make up the difference, adding complexity

THE TRACK. SOME ELEMENTS YOU FEEL MORE THAN YOU HEAR, AND THAT’S A BIG PART OF MIXING TO ME.”

It extends to more than just the backing vocals. On Talk is Cheap there’s a delayed snare panning across the stereo image, and the Fender Rhodes tracks are often moving in subtle ways. It’s all part of helping a mellow record stay exciting. On Blush, a heady, almost dancey track, the drums sit right out on the edge of the image. “It had a different reverb, because it felt like that part was

almost a dream,” said Eric. “I love scene changes in music where for maybe one minute you’re in this other location. I have a very visual sense when I’m mixing, so I’m imagining where the person is standing. It becomes a little different with electronic music. You used to imagine a band in front of you, whereas now there’s a swoosh over here and something else over there. It’s fun to try and mess with people’s expectations, not to be jarring, but to introduce change. AUTOMATING REVERBS IS ABOUT CHANGING SPACE. YOU’RE NOT ALTERING THE ACTUAL REVERB BUT MAYBE YOU’RE CHANGING WHERE THEY’RE STANDING IN RELATION TO YOU, OR USING MORE PRE-DELAY SO THEY’RE MORE IN FRONT OF YOU THAN IN THE BACK OF THE ROOM.

“I use a lot of UAD effects; the Space Echo, Dimension D on vocals, and I always use the EMT140 plate. They’re really good at modelling what would happen if you send a lot of a signal into the unit. So sometimes I’ll send more from the bus and keep the volume low. “My sessions are pretty huge and it’s mainly effects and parallel compression chains. Even if I get a song that doesn’t have a lot of tracks, I send them to a ton of stuff just to try and keep it exciting. “That’s another thing I love about the HDX system. I know if I run out of voices then I’m being over-indulgent. I’ll be doing things like creating a separate reverb channels for two words of a song, either creating a separate delay for one line of a song, or a triplet delay for just one word. “I love doing that. I just mixed a song where I

had four different delay throws, so you never get the same thing twice. It just keeps it interesting. One time it’s a triplet, one time a quarter-note delay, then sometimes it might be a sixteenth-note with a lot of feedback. You can just play with these things, there’s so many possibilities.” On the flip side, for the main vocals, Eric used some longer delays on end sections, but for the most part didn’t want them to be too effected, because he “knew it would lose some of that authenticity”. They still received their fair share of treatment, always going through the Lisson Grove R-124 tube compressor, with some vocals passing through a Neve pre and EQ, all to add more harmonics and intimacy. RHODES TO AN END

The Fender Rhodes is the other main component of Chet Faker’s sound. It’s present on almost every song off the record, but it doesn’t always sound the same. “It was mandated by the song,” said Eric. “Sometimes the Rhodes was the bass. I’ll cut up the track into different pieces so I know if the Rhodes is doing the bass here, then I’ll use all my subs and treat the bass for extra harmonic content. I use three or four parallel compression chains, and have a UAD Voice of God plug-in to get some extra low end coming through on a laptop. “Talk is Cheap doesn’t have a bass, it’s the Rhodes. So there’s a Rhodes track I treated as the melodic part, and another I split out onto a separate channel to EQ and push completely differently.

Left: Just three of Eric’s monitoring options — Avantone Mixcube, Genelec nearfield monitor and Barefoot MicroMain 27s. Then there’s the laptops, iPhones and Bose Sound Dock, which are just as important. Above: The Lisson Grove R-124 compressor in green, and his Bricasti and tube spring reverb above that.

“He mostly played the parts all at once, because he’s a player, and it was all on a real Rhodes. So on Talk is Cheap I took the left-hand part of the Rhodes and put it into Melodyne to get an identical MIDI performance. Then I put synth bass underneath just to add a little bit of low end. “Often people will just give me a whole track and I do a bunch of creative editing and EQ to extract different elements. When people give me a drum loop, sometimes I like to add samples. It’s an interesting experience trying to trigger samples using a drum loop, you have to cut everything into different parts. My friend Chris Shaw recommended Cable Guys’ Volume Shaper for that.” It shows there’s a lot of work that goes into taking a laptop producer’s work and crafting a more detailed expression of the artist’s original vision. But as Rubin and Mardin before him have pointed out, once you have a voice like that, “it’s about providing support, and leading the listener along.” No auto-tuning required. AT 23

REGULARS

STUDIO VIEW DAVE CLARKE Dave Clarke, the Baron of Techno, has a unique approach to his craft. For starters his studio is on a boat in an Amsterdam canal. Secondly, he’s not a synth-head. He’s all about using effects and compression as his sound shaping tools more so than filters and oscillators. His studio is based on a 24-core Hackintosh running Logic Pro, but from there things get a little weird: He takes channels and stems out of an Apogee Symphony into a bank of various hardware compressors then into 24 channels of a summing mixer before being mixed down via a smattering of hardware compression and EQ. Dave calls it a proper ‘hybrid’ setup of in and out of the box. Here we’ve asked Dave to talk us through some of his gear choices.

FAN OF APOGEE SYMPHONY

There’s a Dyson Airblade attached to it because the moment you stick another card inside, it starts overheating. They won’t tell you this. Even if you do put the magnetic swipe on the back, you need to reflect the airflow in a more efficient way. I took the top off the Airblade (on its lowest setting) and attached a duct to it and that feeds air to the back of the Symphony.

HARDWARE COMPRESSION FLAVOURS

I use the Roll Music compressor for bass. I don’t know what it does to bass but it’s very smooth. Sometimes when you use filtered-down bass and you shove it into a compressor you can hear it creaking. But this one just seems to add a bit of peanut butter to the mix and make it very smooth. Not the crunchy peanut butter; smooth. Naturally I use the Distressor for bass drum, but will put more than a bass drum through it. I’ll have it reacting with a bass drum, then maybe a snare or a hi-hat. Sometimes I use the Smart C2 for snare or hi-hats in Crush mode, or send the snare and hi-hats together through the Fatso, because that’s kind of fun as well. The Avalon 747 is for synth sounds or more full synth bass sounds. The Titan is probably more for percussion gain. The API 2500 I often use for vocals or guitars, just because there’s that American sound that can be fun. Otherwise I’ll use the SSL comp [Al Smart C2] on vocals. Previously I had the SSL on the two-mix but found it a little too polite. But it’s great on vocals. AT 24

SUPER STEREO

RMS755

INPUT

747sp

THRESHOLD

-10

- dB +

GAIN +10dB

TSP

ATTACK

TUBE SIGNAL PATH

- dB + EQ

-15

+15

SIDE-CHAIN

COMP IN

RELEASE

160 100 70

SLOW

FAST

200

SLOW

METER

250 300

1k6

600

Hz 1k

700 600

1.2

.15

1.6

2 .1 RELEASE-Sec

DUAL

GAIN

BYPASS

2k5 3k

RATIO IN

MAKE-UP

4k 6k

Hz 10k

0dB

GAIN

+10

OUTPUT

SIGNAL

10 7

3 2 1

decibels

30 24

AVALON

6 + 0 6

CLASS A PROGRAM EQUALIZER

24 L

2 3 4

- dB +

AVALON

EQUALIZER

VA C U U M T U B E

1 + 0 -

COMPRESSOR

CAL GR

400 800 60

HF +15

SC LISTEN

FAST

-15

50 .05 ATTACK-mS

4 8

COMPRESSION

CONTOUR

6 20

.5 .2

RATIO

SIDE-CHAIN THRESHOLD

THRESHOLD 0

COMPRESSOR

4 2

GAIN REDUCTION

LEVEL

DESIGN

CRANESONG AVOCET MONITOR CONTROL

Every time you get a new piece of equipment, it always opens up a weakness somewhere else. After going with the SPL Neos I was missing a big console’s monitoring section. But Dave Hill, f**king genius that he is, has done something special with Avocet. It’s got this wonderful thing called Offset. If I’m monitoring compression before it goes back in the box, I might be hearing something 4dB louder and think it sounds better when in fact it just sounds louder. So you can program in the offset, and suddenly — f**king great idea — you can hear it before compression as it comes out the mixing desk, and then hear it after you’ve gone through all your processing and EQ at the same volume level. You can objectively determine: is this beneficial to me or is it suckering me in?

PARALLEL EQUALIZER

TWO-BUS CHERRY ON TOP

120 Volts Analog 24|2 Mixing Console

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SPL NEOS SUMMING MIXER

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Out of the mixing console into the Focusrite Blue ISA330 into a Bettermaker 232P EQ with full recall via a plug-in into a Clariphonic EQ for a little bit of sizzle, and then into an old Cranesong HEDD. And then from the HEDD back into the box.

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Model 1010

I did have a Neve mixing console. But after a while I realised that a) it sounded pretty shit and b) it was clunky to use — too many button pushes to get the routing right, which annoyed me. Sorry Dave Grohl, but you can’t spell Neve anyway. I was looking at a variety of summing mixers, and the 120V SPL Neos made the most sense and it sounds very good — there’s a lot of transparency and a hell of a lot of headroom. I have 24 channels.

Made in Germany

AT 25

MONITORING: DUAL TO THE DEATH?

For years I swore my undying allegiance to Tannoy monitors. I grew up on Tannoys. To switch camps would be an insult to my father’s memory. I love Tannoy dual concentrics — their tight, controlled sound. But after many years and with mixes getting more and more complicated I realised that Tannoy wasn’t moving ahead. I realised the more complex my reverb tails were becoming, the less I could pick them out within the mix. Gradually I felt my mixes were suffering as a result. With a heavy heart I began looking elswhere. I thought I’d try the Neumanns. I listened to them, they sounded beautiful, but I knew there was some weird trickery going on inside the box I couldn’t trust. Barefoot? I heard those at an SAE workshop in Amsterdam, when I was invited for a tour. I heard them and they sounded bloody awful. Okay, it may have been the room they were in. Then I thought, hang on a minute, you’ve got two bass drivers on their sides facing each other; how does that work with air pressure? After quite a bit of shopping around I finally auditioned a pair of ATCs. I let the guy put them in. He had them sitting on their ends — ‘portrait’ mode — he said the tweeters were happy in either configuration. I listened to various different source material, including Stevie Wonder, and the clav in the Superstition mix started to hurt. Every time he hit it, it was like someone was outside of the ship hull attacking it. Something about the speakers in that configuration didn’t look right. I recall my old flying teacher telling me: “If it looks like shit, it’s going to fly like shit”. Which is a pretty cool thing to say. I was looking at these ATC’s thinking they looked like shit in that configuration. So we turned them on their sides and everything just went ‘click’. It snapped in; everything made sense and tears came to my eyes because it sounded so engaging but not tiring. That’s a really difficult thing to get the speaker to be engaged but not exhausted. To not be flattered, but hear everything. I had them on the Argossy I42 stands, which always worked very well with the Tannoys, but the ATCs are such a heavy speaker I was worried they might topple over when the next canal boat came past. I then put them on the Towersonics. You have the option to fill them with sand or lead. Timing-wise the sound improved by another 10% being on those stands.

EQ PLUG-INS: ON A TRIDENT TIP

Now I’ve gone with a summing mixer, I miss some of the ‘big console’ features, like channel EQ. Which is why I was tempted by the SoftTube Console. Instead I set up a template in Logic which automatically assigns my Waves EQ plug-ins. I’m not a preset/template kind of guy. This is the only template I have and it will choose what EQ module goes to each output. At the moment I’m in a Trident EQ mood.

AT 26

CABLE GUY

I was in Japan visiting friends who gave me some prototype power cable to take home. I actually bought a special case that allowed me to take it on board — there was no way I was going to be separated from this special cable and check it into the aircraft’s hold. This cable was about as thick as a hose pipe wrapped in carbon fibre, made of medical grade silver inside, and apparently the molecules were all lined up going one way… if you believe in that type of thing. The connector was palladium with a glass-filled bead bezel. I didn’t get a wink of sleep for those nine hours back from Tokyo, I was just lying awake thinking about that cable in my studio. It had a 16A IEC at the end of it and I was just imagining it supplying pure electrical goodness to my Furman. That’s the OCD in me. Admittedly I haven’t gone as far as Wendy Carlos and enclosed my whole studio in a Faraday cage, but I’m well on the way!

I HEART LOGIC PRO

I had a break from the studio about 10 years ago and when I came back I thought I’d look into Ableton Live. But it upset me. I didn’t realise why it upset me, and then I realised it upset me because it reminded me of C-Lab Notator. Notator had this two-screen bullshit where you had to constantly flip flop between one screen and the other. I also didn’t like the fact Ableton wouldn’t allow me to commit. It would allow me to keep jamming forever and just keep throwing stuff into the pot. It was fun but I wouldn’t get anything finished. I re-learned Logic after years of working in Cubase. It was very familiar but at the time it was so unreliable. Logic 8 and Logic 9, it just wasn’t fun at all, and the 32-bit bridge would constantly f**king upset the whole applecart — you’d go out for lunch, and you’d come back from lunch and it’s crashed for no reason. But when Logic 10 came out, I was able to make the full leap to 64-bit. And you can now take its stability for granted. Now, 10.1 finally has the Plug-in Manager and I now feel it is an absolutely complete and reliable piece of software, a joy to work in finally! It’s my friend. I love it.

PLUGS: MAKING WAVES

I have about 780 plug-ins [including the Dave Clarke signature suite – Ed.]. I’ve been using Waves Mercury for a very long time, since back in the days when it took three minutes to load up a single plug-in. Over the years I’d be running plugs on UAD from Firewire boxes to internal cards and right now I have a 24-core Hackintosh, with two PCI cards in there so I can run a shitload of plug-ins in one go. For a long time I was a big fan of Plugin Alliance, but the update routine is such a palaver. But I’ll use those and a few others like the BBE plugs, which are good fun to have.

“

T he connector was palladium with a glassfilled bead bezel. I didn’t get a wink of sleep for those nine hours back from Tokyo, I was just lying awake … imagining it supplying pure electrical goodness to my Furman

”

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TUTORIAL

AT combs through your measurements to understand why flat-lining a PA is probably a bad idea. Tutorial: Ewan McDonald

You’ve been there. You might have even done that. We’ve all got anecdotes of system techs and FOH engineers pulling out their measurement system to tune a PA, massaging the trace line to a ruler flat response, then pronouncing to all who come hither that the PA is now ‘linear’. Good in theory, right? But now everyone in ear shot is wondering why the PA sounds a lot worse than it did when first turned on. If this misalignment between ‘flat looks’ and ‘good sound’ has ever bothered you, or you’ve been struggling to get consistently good results when using your measurement system, then read on. FLAT-LINED & LIFELESS

Common reactions to PAs that look flat are they sound ‘lifeless’, ‘harsh’, or are ‘just plain bad’. But, believe it or not, there’s actually a very strong correlation between flat loudspeaker response AT 28

curves and favourable listening preferences — though only in very controlled environments like anechoic chambers with a single reference loudspeaker. The problem is, we don’t work in these types of acoustically sterile environments. The other reaction I often hear when someone liberally applies EQ to flatten a trace is that too much EQ affects phase. True. But there’s a minimum phase ‘window’ we can safely play in when tuning a PA. In fact, IF THE TUNING ISSUE IS ‘MINIMUM PHASE’, THE PHASE RESPONSE IS ACTUALLY FIXED BY IMPLEMENTING A STANDARD EQ FILTER PROPERLY.

The difficulty is in determining which parts in the measurement fall under this minimum phase jurisdiction, and hence can be fixed with EQ. We’ll get to that down the track. So if we can’t simply flatten the trace, what is the solution? In this series of articles, we’ll delve into how to use your measurement system to make

more effective and accurate adjustments, and positively impact that end goal of optimising your PA system. Let’s face it, if it was as easy as hitting a pre-defined curve — flat or otherwise — auto-EQ systems would actually work and a lighting guy could tune the PA for you. The number one reason why flat responses don’t often sound good is that a lot of what we see in a measurement trace is due to factors that cannot be fixed with EQ. So as you add EQ to make the line look better, you may be unnecessarily colouring the direct response from the speaker and creating more problems than you’re solving. OFF TARGET SHOOT OUT

In my past life working for a speaker manufacturer, I was privy to many loudspeaker shootouts where two or more speakers were compared side-byside. Under very controlled circumstances a single

“

Diagnosis 101 – Initial Measurement 9 6

Using a measurement system without knowledge of audio physics is like a chef trying to read an X-ray

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consultant or engineer would tune each PA to be almost completely identical in the frequency domain. And I can tell you first hand that every single time this balancing act was performed the speakers still sounded different — and more often than not, the differences were not subtle. WHAT YOU SEE ON A MEASUREMENT SCREEN IS THE CULMINATION OF A HUGE ARRAY OF FACTORS IMPACTING ON AN INCREDIBLY SMALL SAMPLE SIZE — THE POINT IN SPACE AT THE TIP OF THE MICROPHONE. Move the microphone a small distance and you may see quite a different picture. Then subtract from the melting pot all the information we see in a transfer function trace that our ears almost completely ignore — hint: there’s plenty. And conversely, add all the audible cues our ears definitely take note of that won’t show up in the transfer function — distortion and transient response are two examples — and we’re left with a difficult problem to solve. Here’s just a small glimpse into what’s actually going on behind what we see on the screen:

Loudspeaker Response Crossovers Loudspeaker Interactions Reflections Reverberation Room Modes Microphone Response Environmental Factors

The Engineer’s Ability to Interpret the Data

Measurement Hardware

”

DIAGNOSIS 101

Let’s start by having a look at the above measurement, and asking the question: What would you do to this trace? Would you: A) Start applying EQ to fix those big dips and peaks in the low mids. B) Begin pulling the PA apart — something is obviously broken or out of phase. C) Leave it alone, it’s probably a ground reflection. D) Nothing. How could I? I know nothing about what the measurement was taken of, and how, when, or where it was taken. The only real answer is D, but if you answered C, then you’re halfway there. If you answered A or B — then you definitely need to read on! TO GET BETTER AT MEASURING PAS THE FIRST THING YOU’LL NEED TO DO IS RE-VISIT YOUR AUDIO THEORY. Now don’t feel bad if your theory

or measurement skills are lacking. I had a measurement system set up in front of big systems for a long time before I really knew what the heck I was doing with it, or more importantly, before I properly understood the complex audio world around me. What I soon discovered was that using a measurement system without knowledge of audio physics is like a chef trying to read an X-ray; you may be able to see that something looks broken or weird, but you’ll have no idea what you should do to fix it. And if you do try and fix it, you’re probably just going to make matters worse. Believe me, I’ve worn that chef ’s hat more than once!

COMBING THROUGH IT

If you’re wondering where to start, a good place is sound source interactions and comb filters. Understanding what happens when sound sources interact with other sound sources, or with reflective surfaces, is absolutely crucial — not only from a system design and alignment perspective but a measurement perspective. Why? AT 29

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TRACING BACK If you intend to learn anything from your measurements later on, you must develop an appropriate naming convention — preferably before your first measurement. Firstly, establish your own shorthand for each position (it could be G for ‘mic on ground’ or S for ‘mic on stand’) and which speaker you’re measuring (L/R, Dly, etc). You also want to include a numbering system to indicate the distance from the speaker (or position front/ back if you don’t have time to measure the distance) and note whether the system is pre- or post-EQ. It allows you to look over the measurements offsite. That way if a system really gave you a hard time, you can go back and check your measurements to try and get a better idea of what was going on.

BECAUSE THE VAST MAJORITY OF THESE INTERACTIONS THAT SHOW UP AS FREQUENCY RESPONSE ISSUES CAN’T BE FIXED WITH EQ, SO WHEN YOU SEE THEM ON THE SCREEN YOU’LL NEED TO KNOW WHEN TO LEAVE THEM ALONE.

Now most of you should know what a comb filter is, but when was the last time you really thought about it while looking at a measurement trace? The most common and usually the most severe comb filter is the one from the ground reflection, so we’ll focus on that first. The picture you judged earlier was a PA that has a reputation for excellent low-mid response, but with a severe ground reflection showing up in the measurement — any attempt to make that trace look better in the low mids would have been a mistake. AT 30

1.5m

THE GREAT OUTDOORS

Now the fun part begins. Get out your measuring kit and let’s start with a couple of experiments. While this might sound like a bit of work, doing your own controlled experiments will teach you more in one afternoon than months of trying to learn on gigs, installations, or reading text books. Your measurement system is an invaluable tool in not only aligning and tuning systems but helping you learn. Grab yourself a speaker. What type is not important, but if possible, use the same speaker you’ll be measuring in the field to start developing an idea of what that speaker should ‘look’ like. On a calm day, take one speaker outside and place it on concrete, or any other hard surface. Other hard reflective surfaces nearby are going to affect your results so put it as far away from walls and cars as the space allows. In a perfect world you’d be doing this in an open field, so the more space the better. Now if you’re using a relatively small speaker, put your speaker on the ground and the mic about two metres away. Place something under the back of the mic, so the actual capsule is as close to the ground as you can get while still pointing at the speaker. Then tilt the cabinet down a little so the mic is on axis. This is closer than you’d measure in the real world, but the aim is to minimise the effect of any environmental and acoustic factors for the sake of the experiment. What you also need to note is that when you’re measuring too close to a speaker, the interaction between the drivers within the cabinet can heavily influence the measurement — so if something looks dramatically weird, move the mic back a little and see if it smooths out. Now measure using the transfer function without too much smoothing. If you use too much

smoothing you won’t properly see the comb filter; if you use too little, things can get confusing. I’ve found 1/12th is a good place to start, but 1/6th may give you a picture that’s a little easier to read. WIND CAN WREAK HAVOC ON YOUR MEASUREMENTS, SO USE LONG AVERAGES OF FOUR OR MORE SECONDS, AND MEASURE A FEW TIMES IN EACH POSITION. If the measurement doesn’t change then you’re good to go. If it does, it’s best to try another day.

COMBING IN THE MIRROR 1 Once you have the first ‘ground’ trace stored and labelled, place the cabinet at a height of 1.5m and the mic on a stand at a height of 1.5m, and re-measure at the same distance. You should notice two things: the first is that the ground plane measurement is 6dB higher than the mic stand measurement; the second is that the mic stand measurement now has some big peaks and dips in the Low Frequency (LF) region. First, let’s deal with the level change. No, somebody didn’t bump your preamp gain. This +6dB ‘for nothing’ is the same effect you get with a boundary mic where the direct sound and the reflected sound from the surface arrive essentially at the same time (and hence almost perfectly in phase at all frequencies). Because the mic is ‘on’ the surface, it’s the equivalent of placing the mic directly in the middle of two sources where you’d see a 6dB increase in pressure. 2 Think it should be 3dB? Well, I guess you just learnt something! This brings us to our next point about direct and reflected energy: 3 whenever you have a sound source and a reflective surface, the sound field acts the same as it would if you had a ‘mirror image’ of that source on the other side of the surface, and the surface was removed (albeit with the effects of

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2 Levels in decibels (dB) and their different factors 30

Loudness

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Voltage 12 9 6

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Factor (log) Sound Intensity Level: Twice the Power registers as a 3dB increase. Sound Pressure Level: Twice the Voltage registers as a 6dB increase. Psycho-Acoustic: Twice the Loudness registers as roughly a 10dB increase.

surface absorption applied to the mirrored image). Understanding this virtual mirror image (or image source modelling, as it’s known in acoustic simulation circles) is handy, because once you know how two sources interact you can then apply the same knowledge to a single sound source hitting a reflective surface. An acoustic two-forone deal. If you want some proof of this virtual mirror image phenomenon (and have the space to do it) grab an additional speaker of the same type and place both on the ground three metres apart. Place your mic directly in front of one of them (ie. on axis of one speaker only, not between the two) at the same distance away as the first experiment. The dips and peaks should appear in the same spot as your previous measurement with the ground reflection. Remember: No surface is perfectly reflective, and loudspeaker coverage patterns are not omnidirectional at all frequencies. This means that late arriving energy does not have the same frequency response or amplitude as the original signal and these combs may become more or less severe based on the situation, though the spacing of the peaks and dips should remain the same. PEAKS & DIPS

On to the peaks and dips. We know a comb filter is created by two identical sound sources arriving at a single point at different times. Here we have the direct sound from the loudspeaker arriving at the microphone position first, and the reflected AT 32

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sound from the ground arriving late. You should be able see the discrete late arrival by looking at the impulse response in your measurement system. 4 THE EASIEST WAY TO FIND OUT WHERE THE COMBS OCCUR IS BY MEASURING THE TIME DIFFERENCE BETWEEN THE PEAKS OF THE IMPULSE RESPONSE, THEN USING THE FORMULA 1/(2 X TIME DIFFERENCE BETWEEN ARRIVALS) TO FIND THE POSITION OF THE FIRST DIP. THINK

What happens when two perfect sine waves are out of phase? They cancel each other out completely. But when they’re perfectly in phase you can only ever gain 6dB, so the dips are always going to be easier to see than the peaks. Let’s say the time difference is 5ms. We first translate that into seconds — i.e. 0.005s — then multiply it by two, to get 0.01s, then divide one by 0.01. The result should equal 100Hz. This is the frequency of the first dip you should see. There should then be a corresponding boost at 200Hz, a dip at 300Hz, a boost at 400Hz, another dip at 500Hz, and on it goes. When you think about this, the time period of 100Hz is 0.01 sec,

BACK TO WHAT WE KNOW ABOUT SINE WAVES.

so an arrival of 0.05 is completely out of phase at 100Hz and completely in phase at 200Hz, just one cycle late. Now, since frequency lives in a logarithmic world and a comb filter appears in a linear fashion, as we move higher in frequency the peaks and dips get so close together that they effectively cancel each other out, and smoothing further hides the ripples. So usually once you get past the first three or so dips the comb gets very hard to see clearly. This is compounded by the fact that higher frequency information is often absorbed more effectively than lower frequencies, and higher frequencies are more directional (ie. if the speaker is facing straight forward, high frequencies don’t radiate into the ground as much as lower frequencies). In a nutshell, the comb effect becomes less severe as we move upwards in frequency. So take the number of the first dip we calculated, multiply it by three to find the second dip, and five to find the third dip. If you find these three dips in succession, there’s a good chance you’ve found yourself a comb filter.

Stand vs Ground at 2m

5 TRACING ON PAPER

Back to the original, single-speaker setup. If you can’t wait until you have the time to do this experiment for yourself, then here’s a couple of measurements I took using a great sounding twoway speaker. 5 You can clearly see the ground plane measurement (6dB higher like we discussed) and then the measurement with the ground reflection. What you can see is that the bottom end appears to have high-passed and there’s now a large boost around 220Hz, then a dip just above 330Hz, then a peak around 440Hz and another dip at 550Hz. Now the first dip of the comb filter is actually centred around 110Hz, but the High Pass Filter (HPF) in the cabinet makes it a little hard to see, and it just appears like the HPF has been moved up. If you do the math, this is exactly where these comb filters should be occurring. 6 Here’s another one, this time with the same speaker and mic on the stands, but measured at five metres. For a 5m distance and 1.5m speaker/ mic height, the initial comb should be very close to 200Hz, with the third and fifth dips around 600Hz and 1kHz, respectively. Now some of you may ask, “Doesn’t the Time window get rid of late arriving energy in the measurement?” Well, unfortunately, this only works for higher frequencies. Once you move to lower frequencies (where these interactions have the most impact) the measurement system needs a long Time window to get enough resolution in the frequency domain. This means most of the reflections that do major damage still arrive within the Time window. Those of you that have ever worked with a single Time window (unlike the multi-Time window systems mainly used these days) will be very aware of this issue. The ground reflection is no different to any other reflection, just that it’s the only one that will show up every time you are measuring when the mic is raised off the ground. The theory is exactly the same for side walls, rear walls, ceilings, large objects, or any other surface that reflects the sound source back towards your microphone. So once you’ve mastered picking up on that ground reflection, stay outside and start setting up your speaker next to walls and other reflective surfaces and try to pick them up in the measurement too. First try the mic on the floor so the wall reflection is prominent, then take the mic off the ground so you’ve got both the wall and the ground reflection… Now take it inside where you not only have reflections, but room modes and reverberation to worry about! Getting complicated? Damn right! But hopefully this is starting to give you a picture of how much more there is to measurement than simply flattening the squiggly line. In the next few articles, we’ll start looking at getting to know the speaker you are measuring and why this is important. As well as some tricks and tips on measuring in a reverberant environment; using that dreaded phase trace; and finally, putting it all together including tips on measurement settings, mic placement and EQ practices.

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MEASUREMENT MICS MEASUREMENT SOFTWARE

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QUICK MIX

The

with

Tim Whitten Interview: Neil Gray

Who have you been touring with recently? Low Lux, Lanie Lane, Philadelphia Grand Jury, Paul Dempsey, Deep Sea Arcade, and The Church. Who are some other bands you’ve worked with? The Necks, Brian Jonestown Massacre, Canyons, and Boy And Bear. How long have you been doing live sound? For 34 years, even since a friend insisted I do sound for his band. What is your favourite console and why? Ideally an analogue console with lots of outboard compression but I love the control a digital console gives you. For small venues with less than ideal PAs, I would prefer a digital console for the control and extra processing options. Favourite microphone or any other piece of kit? My favourite mic is the one that suits the situation the best; the instrument and situation will determine what mic works better than another. But if I had to do a gig with one type of mic on all instruments and voice I could survive with a Shure SM58 or SM57. I can’t really live without a Smart C2 compressor on the mix bus. For me it’s the most helpful bit of outboard apart from a vocal compressor, a delay and a reverb. Most memorable gig or career highlight? Mixing Morrissey’s first show after the Smiths broke up was very memorable for the chaos. Mixing Alice Cooper at Reading festival in the ’80s for such a huge audience was a very surreal experience, because I was mixing songs I’d played on my bedroom record player as a kid. The Underground Lovers at the Annandale during the ’90s was always a sonic event. All the Necks shows. It was amazing how three people can improvise and never repeat the same piece of music, yet the music be so transcendent of time and sonic knowns. Joanna Newsom for the sheer beauty of her music. What are three mix techniques you regularly employ? Understanding what it is about the artist’s music that might make people want to listen. Limiting spill into mics if you are working with a less than ideal stage sound. I try to concentrate on the mix balance first, and then the tone. It’s important for the music to make sense, that you hear all the instruments balanced relative to each other. Also, walking away from the mix position in the early part of the set if I can bear to take my hands off the faders — I keep telling myself I should do this more.

AT 34

“

Mixing Alice Cooper was surreal. I was mixing songs I’d played on my bedroom record player as a kid

”

Game changing gear from the last 15 years? Line arrays and the processing power to fine tune a setup, IEMs, digital consoles, and laptop control options. How have your working methods changed over the past 15 years? System setup can now be more scientific but I still need to trust my ears after all the screwdriver work is done. Any tips/words of wisdom for someone starting out? It’s good to ask questions and read lots. I used to sneak into sound checks and try to pick up a few tips, or if it was a touring PA, I’d offer to help load in and help rig the system. Learn to listen to sound and relate it to music. Try to be friendly and understanding and find a band to hang out with and help them carry gear. They’ll need someone to mix them one day, and a lot of the time it’s about being in the right place at the right time. It’s pretty much a people person industry so you need to be able to get along with everyone.

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REGULARS

PC Audio DIY advice that will bring a smile to your face and silence to your studio. Column: Martin Walker

When I first started writing about using desktop PCs for audio nearly 20 years ago, we musicians had major problems with noise — the whirs, clangs, whooshes and squeaks emitted by a collection of cooling fans, hard drives and tinplate casings… and it’s still a concern. Many musicians record vocals and acoustic instruments in the same room as their PCs, and even if your music is electronica created entirely in-the-box, you’re likely to make slightly different mixdown decisions if you can hear the steady roar of cooling fans in the background. Thankfully the multimedia world at large has since become attached to the idea of a quiet living room PC, and as always, mainstream interest inevitably results in more competitive prices. So nowadays, you don’t have to resort to cutting a cable hole in the wall to site your computer in the next room, or place it in an expensive custom-designed soundproof cupboard, as most specialist audio PC dealers offer quiet and even totally silent PCs (with no moving parts) in their range. Nevertheless, there are still many musicians out there who wish their computers were a little quieter. So, first of all, take a little time to clean the inside of your machine. I recently whipped off my computer’s side panel and was shocked to see just how much muck had been sucked in via the front intake cooling fans, even though my studio remains a resolutely non-smoking environment. After powering down and unplugging the mains lead for safety’s sake, a few minutes with a camel hair brush inside the machine while holding a vacuum cleaner attachment nearby to extract the debris left me with clean air filters at the front, sparkling fan blades, and heatsinks and circuit boards that were shiny rather than coated in a patina of grey dust. Blow me (pun intended), when I powered it back up my computer was noticeably quieter already, simply because air could pass through more easily! Next stop with a noisy computer is to look for and deal with the weakest link in the chain, so have a crawl around your machine to see what component is producing the loudest noise. For some people this may be the cooling fan in the power supply, especially if you’ve purchased a AT 36

budget PC, but this is easily remedied by replacing it with a more capable PSU containing a larger and slower-running fan (120mm is typical, and will be a lot quieter than any smaller fan moving the same amount of air). If you want reliability then don’t skimp here; buy a power supply from a well-known brand (I have a soft spot for Seasonic) with enough capacity to run everything. If you’re unsure about the required capacity, pay a visit to www.extreme. outervision.com/psucalculatorlite.jsp and add up the requirements of your main components. Don’t be surprised if the total is a lot lower than you expect — my overclocked quad-core machine consumes less than 300W and is perfectly happy with a manly 500W PSU. If the noisiest item is your CPU cooling fan, then replacing the combined heatsink/fan assembly with one utilising a larger 140mm fan is an option, although not as straightforward as a PSU swap, especially if you’ve not dealt before with the likes of thermal paste (which ensures the best heat transfer between processor and heatsink). A handy guide can be found at www.wikihow.com/ Apply-Thermal-Paste, but if you need to consider such a major replacement I’d advise seeing if your motherboard supports a faster CPU so you can invest in a new combined CPU/heatsink/cooler and throw a system performance boost into the bargain. One of the easiest ways to lower acoustic noise levels is to replace older mechanical hard drives with one or more modern Solid State Drives, which have no moving parts and hence make no noise at all. SSDs will also provide significantly faster Windows boot-up and application launch times, although they are unlikely to boost audio or sample streaming performance compared with traditional hard drives (whose sustained transfer rates can already support 100+ audio tracks and hundreds of sample voices without breaking into a sweat). If you’ve got enough money to replace all your mechanical hard drives with larger SSD models then you’ll certainly end up with an extremely quiet PC, but if not I can recommend buying one SSD for Windows, but sticking with mechanical hard drives for your audio and sample data. However, rather than bolting the latter into drive bays inside

your case, or even using rubber grommets, use elastic suspension instead. This now ancient thread from the Silent PC Review forum entitled ‘HDD vibration & noise reducing methods – ranked’ (www.silentpcreview.com/forums/viewtopic. php?t=8240) offers classic advice for the DIY’er that stands the test of time extremely well. If you’re considering building yourself an entirely new PC then choosing a suitable case is paramount to achieving low acoustic noise levels. I’d recommend one that features a hinged front door that completely covers any DVD drives, since this not only prevents any remaining mechanical noise escaping from the inside of the case, but will also quieten the DVD drive when it’s being accessed. I’m personally a fan of the Fractal Design Define series of cases, which incorporate high density noise-reducing material in their main panel construction, a three-speed fan controller that on its slowest setting is well nigh silent, yet modestly priced. My current PC is built around such a case and contains a quad-core CPU overclocked to 4.4GHz, 8GB of RAM, a modest graphics card (only gamers require faster), one SSD and two mechanical hard drives, yet is almost silent. In fact, a buzzing transformer in my racked power amp became more annoying, that is until I eradicated that buzz by whipping off the amp’s top panel and inserting some neoprene strips between it and the mains transformer. Then my two mechanical hard drives became the loudest components in the studio, whereupon adding elastic suspension to them returned my computer to ‘is it on yet?’ status. Silence is golden, but may be cheaper than you think!

REGULARS

Apple Notes Commanding deep features in Logic 10.1 Column: Anthony Garvin

A cursory scroll through Logic 10.1’s What’s New feature list shows a significant update to the DAW — sound library additions, plug-in enhancements, new editing, automation, and mixer features, and more. But since updating and digging deeper into the list, I’ve found it’s actually an enormous update with over 300 new features, bug fixes and other improvements. If like me, your Logic X journey has been a little bumpy so far, with intermittently frustrating bugs or quirks. Just a couple of days with the update served to allay many of my concerns over the previous version’s foibles. I still stumbled over a few unexpected surprises, but I’ll get to them later. So, aside from the new features listed anywhere you care to search for them, here are a few of my favourites you may have missed. Controlling Plug-ins via iPad — When I first read about this feature, I thought all my iPadDAW-Controller dreams had come true. I’ve written at least twice before in this column about the iPad being seriously under-utilised for studio production tasks — due to a lack of appropriate and forward-thinking apps. So when I heard Plugin View in Logic Remote would allow users to tweak Logic and AU plug-ins via the iPad directly, I got excited. As it turns out, the feature isn’t all I’d hoped for, but still useful. What it does is allow you to tweak or automate all your plug-in parameters from the iPad, which obviously means you don’t have to rely on your mouse for such duties. Parameters don’t have to be mapped to Smart Controls anymore to be accessible via Logic Remote, and all parameters in the plug-in are available via this view — with 24 parameters displayed per page and the ability to swipe between pages. The caveat is it’s akin to using the plug-in’s Controls view; the iPad won’t render the GUIs your favourite plug-in designers laboured over — neither for native Logic nor AU plug-ins — just a standardised layout. It will take a little while to learn where your favourite parameters are, but once you do, it will make effects automation more tactile in a way that nothing else reproduces without additional setup time. However, recording multiple parameter changes

at the same time (via multi-touch on the iPad) wasn’t reliable — hopefully this is improved soon. Repeat (aka Duplicate) — ‘Repeat Regions’ is now just ‘Repeat’, allowing you to repeat any selected region quickly by key command (Command+R by default) — a feature I’ve often missed when going back to Logic after using another DAW. If you prefer the old way of choosing the number of repeats via the dialogue box, that is now called ‘Repeat Multiple…’, also assignable to a key command. By the way, if you’ve updated an existing installation, you’ll need to assign the ‘Repeat’ key command yourself, but if you’ve done a brand new install (i.e. not an update from a previous version), it will default to Command+R. Multiple Drummer Tracks — Strange there was ever this limitation, but now you can load as many drummers as you wish. Well, at least 10 anyway — I tried recreating Hans Zimmer’s Batman drum troupe but was a few drummer’s short. Zoom to Fit Selection — A minor one, but this has been a long-time annoyance of mine. If you are using this key command (Z by default) to zoom out to see the whole project, it should now include looped regions right to the end of the project, and not just the parent region… However, after trying this on a few different projects, I’ve only been able to make it work once. Toggle Musical Grid — This is an assignable key command that allows you to toggle the ruler between bars and beats or minutes and seconds — something that has not been quickly flippable since Apple made changes to the ruler functionality in Logic 9. Compressor Histogram — If you find yourself ever trying to explain to anyone what a compressor does, then this might help. Similar to Ableton Live Compressor’s Activity view, the histogram displays a graph of the gain reduction taking place over time. Audio Device Controls — When recording audio into Logic with a compatible audio interface connected, you can now control the physical input gain/phase/HPF/etc. on the interface via Logic’s Channel Strips. Currently the Apogee ‘pro’ models (from the Duet upwards) support this feature, and as the feature is part of CoreAudio, let’s hope that

we see more manufacturers supporting it soon. In my tests Logic wouldn’t save or recall audio device settings from project to project, though according to Apogee and Apple this should work fine. Keep your eye out for an Apogee software update coming soon. Converting Interleaved Files to Individual Mono — There’s a listed new feature that states “Logic can now import multichannel interleaved audio files and convert them to individual mono files” — which initially grabbed my attention, as Logic lacks any easy way of converting/splitting various interleaved files to separate mono files. Upon further digging, this feature is for interleaved wav files with more than two channels, so it’s not a new solution for stereo files, however it’s useful for those working with field recorders. Dubstep Drummer — It’s nice to see that Logic is trying to stay relevant with dance music trends, even if it is a few years too late. System Requirements — On a final note, this is the end of the line for Mountain Lion, as Logic Pro 10.1 supports Mavericks and Yosemite only. You also have to be running iOS8 to access the latest version (1.2) of Logic Remote. You can check out all the details on the 10.1 update here: http://support.apple.com/en-au/HT203718

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REVIEW

APOGEE ENSEMBLE THUNDERBOLT How was Apogee going to make its audio interfaces any better? By doing what it always does; ushering in the next generation of technology. Review: Brad Watts

NEED TO KNOW

Apogee is certainly not a company to sit about on its collective hands, or rest on its laurels. That said, with the quality of Apogee product already in the market, it could feasibly do so for a quite a while to come. Yet that doesn’t curtail the company’s relentless improvement of its digital audio offerings. Apogee has been at the forefront of digital audio conversion since the very beginnings of the format, having been founded by Australian Bruce Jackson back in 1985, along with Christof Heidelberger and Soundcraft USA president, Betty Bennett. For many years, Apogee DACs and ADCs were the bastion of the professional sphere, with devices aimed at the upper echelon of the recording and mastering world.

PRICE $3499 CONTACT Sound Distribution: (02) 8007 3327 or info@ sounddistribution.com.au

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PROS Plenty of I/O Minuscule latency Great mic preamps No need for monitoring or talkback hardware Re-amping built-in Sublime sonics

Yet from 2007, with the release of an audio interface aimed at the project studio market, the Firewire-based Ensemble interface brought top-shelf Apogee audio conversion to a far wider audience. At the time I had the pleasure of auditioning and reviewing the original Ensemble, and as some may recall, opened a mini can of worms when in 2007 I described the Ensemble’s D/A conversion as “cuddly”. Well, let me assure you, the newly-minted Ensemble Thunderbolt is even cuddlier; it will be met by all and sundry with wide open arms. It’s now eight years down the track, and Apogee’s continuation of the Ensemble heritage is quite a remarkable evolution. Gone is the

CONS A dim display? Who cares!

SUMMARY Apogee’s Ensemble has always been a workhorse. But the new version — with Thunderbolt, a new DAC, JFET DIs, reamping outputs, talkback and multi-monitoring outputs — is a studio centrepiece.

There’s a lot of great features packed into the front and back panels of the new Ensemble including two reamping outputs to go with the pair of JFET DIs; eight Apogee preamps with 75dB of gain and all the trimmings; S/MUX expansion; and not one, but two (hallelujah!) Thunderbolt 2 ports.

original unit’s Firewire connectivity, supplanted by the wider, faster, and vastly more versatile Thunderbolt connectivity (Thunderbolt 2 provides 20Gb/s compared to Firewire 400’s paltry 0.4Gb/s — that’s 50 times the bandwidth). This fact alone leaves the Ensemble steadfastly within the realm of Apple computers — a strategy Apogee is more than comfortable with, having renounced development for Microsoft operating systems during early 2009. Physically, the new Ensemble Thunderbolt is one of the sturdiest pieces of audio gear you’ll come across. The casing is constructed of 2mm thick steel. There’s not a scrap of aluminium to be seen here. Equally reassuring is the fact all the I/O ports are firmly bolted or screwed into the chassis, so there’ll be no fear of bending and damaging connectors held to the board by way of solder connection alone. Plus, in somewhat of a recent departure for Apogee, the Ensemble is black. Perhaps this is, in part, consistent with the rebadging game going on between Apogee and Avid (such as the Avid Duet and Quartet), or maybe it’s simply time for a new look and a move away from aping the old Mac Pro/Apple aesthetic. Either way, the blacker the better as far as I’m concerned. EYE OH, EYE OH

The Ensemble’s I/O quotient is bountiful, and includes eight of Apogee’s high gain (75dB) mic preamps. The preamps incorporate Apogee’s Soft Limit clipping protection and saturation, 48V phantom power, and high-pass filters. All these features are switchable via software, or you can toggle through the input controls after holding the input button down for three seconds. And for the record, phantom power is switchable for individual channels. Connection to the majority of the I/O is via the rear, with the first four inputs being combo inputs for both XLR or jacks. The remaining four mic inputs are dedicated XLR inputs. Inputs 1 and 2 include inserts. These are set up with the useful arrangement of separate jacks for in and out as opposed to a single TRS jack. Outputs 1 and 2 are presented as balanced jack outputs with the remaining eight balanced analogue outputs presented via a 25-pin D-sub connector.

A further 16 I/O points are available via four TosLink/ADAT optical connectors, which can be configured for S/PDIF or S/MUX for 96k operation (fine for I/O expansion, or indeed, incorporating digital effects or instruments). Alongside the optical ports are coaxial S/PDIF in and out, wordclock in and out, and two Thunderbolt 2 ports. All up you’re looking at 30 x 34 I/O points (including the two quite loud and independently-controlled headphone outputs). Access to the entire I/O quotient will, of course, require additional ADAT interfaces, which leads me to wonder if Apogee has plans for a single rack unit, eight channel ADAT I/O unit (please oh pretty please, Apogee?).

“

Top marks Apogee – a creative gold mine built right into the interface

”

OH(m) THE IMPEDANCE!

Out front are two additional high impedance instrument inputs. These incorporate Class A JFET circuitry for more pleasing harmonic character when recording guitar and bass. JFET circuits are extremely high impedance and are voltagecontrolled rather than current-controlled like a solid-state transistor. Consequently a JFET circuit offers some attributes which are similar to vacuum tubes in the form of those additional harmonics we all love on guitars. What’s exciting about this section are the two additional instrument outputs. These are provided for re-amping duties, or indeed, strapping high impedance stomp-box effects across a track. Top marks Apogee – a creative gold mine built right into the interface.

I’ve seen many an audio interface, and nothing else can pull this trick with the correct impedance matching. Hence the market in outboard ‘reamping’ devices to match line level interface I/ Os with amplifiers and instruments. I’d recently re-built my pedalboard with a pile of 1980-ish stompboxes and had a grand ol’ time running all sorts through pedals. Drums through a Morley wah, vocals through a RAT pedal. Brilliant. An utterly exceptional feature thanks Apogee. FUTURE’S BRIGHT?

Visual feedback such as metering and headphone output levels is delivered via OLED displays, or of course, via the downloadable Maestro 2 control panel software. Speaking of software, you’ll need to be running a minimum of version 10.9.3 of OSX (aka Mavericks). But more on that in a moment while we peruse the Ensemble’s front panel. OLEDs certainly look set to be the next regime in metering. And while most will agree they look fantastic in their 21st-century sci-fi splendour, I can’t help feel they’re a tad dim. Sure, the higher resolution is wonderful, but I’d prefer to see them from a distance in a lit room. I’ve yet to see a device with OLEDs that includes a brightness control and the Ensemble is no exception. Anyhow, putting that single gripe aside, straddling either side of the OLEDs are large detented potentiometers. The left side pot adjusts the selected input’s gain, and the right side pot adjusts monitoring level. A push on the right pot will mute or un-mute the main outputs, and a push on the left pot scrolls through the 10 primary input channels (1 through to 8 and the two guitar channels). To the left of the input level pot are dedicated backlit buttons for quickly selecting input channels for adjustment. Off to the right are four assignable buttons which can be assigned via the Maestro 2 application for dozens of functions. These can include options such as clearing meter peaks, through to toggling the guitar channel outputs from the DAW software or the guitar inputs. What’s interesting is the ability to assign a button to engage talkback via the built-in front panel microphone. The talkback destination can be set to go to either or both of the headphone outputs, AT 39

and Outputs 9–10. That’s right, the Ensemble Thunderbolt entirely negates the need for an external monitoring and speaker selection device. This ensures you’re hearing every skerrick of sonic goodness from the Ensemble without sullying the fidelity with an inferior monitoring unit. You can even assign outputs to three sets of monitors and choose to use an external mic for talkback. Absolute gold. Interestingly, the built-in talkback mic can also be routed to a channel and therefore its output is recordable (handy in a creative sense perhaps, but definitely useful for keeping Hansard records of each and every session). THE ROUND TRIP

For decades, one of the more unfortunate aspects of CPU-based ‘native’ recording interfaces has been the issue of latency — ie. the delay experienced between the source material and the monitoring of that signal from a DAW via the studio monitors. When recording live performances this isn’t a huge issue, however, when it comes time to add overdubs, every millisecond of processing delay adds up. If the vocalist is hearing their voice some milliseconds after they sing, it can completely destroy a performance. Years ago there were a couple of ways to get around this issue: either have your interface set for ‘direct monitoring’ (whereby the performer’s signal was monitored directly from the interface rather then via the DAW and its inherent processing lag), then adjust your recorded track to put it back in sync with the multi-tracked parts; or spend AT 40

thousands of dollars on DSP-assisted recording interfaces such as Digidesign/Avid Pro Tools hardware. For the majority, the latter option was fiscally untenable, and direct monitoring and adjustment was the only choice. Since then, CPU processing speeds have increased dramatically, along with the number of processing cores in a CPU, and wider processing buses such as PCIe, and now Thunderbolt. This has allowed latency figures to lower considerably. But go back 10 years and it was common to experience 5–8ms of delay between performance and monitoring. Apogee has made inroads into alleviating the latency headache, and made huge headway with its Symphony PCI and PCIe cardbased systems. Apogee’s card-based platforms, combined with multi-core, high-speed processors do make latency an almost negligible consideration. Where Apogee’s technology gains its edge is when running at higher resolution sample rates such as 88.2 and 96k. The new Ensemble with its Thunderbolt 2 connectivity competes side-by-side with DSPassisted systems. At a low buffer setting of 32 samples in Logic ProX and running at 96k, Logic reported a round trip delay of a paltry 1.1 milliseconds. That’s a minuscule delay, and one which would never leave a performer feeling as if their monitoring could be slightly out of whack when listening via the DAW rather than directly monitoring from the interface. What this means for the engineer/operator is there’s no longer the need to reorganise patch and routing setups between

tracking and overdubbing situations. It’s firstly a time-saver, and secondly, there’s less of a tendency to find tracks out of alignment when it comes time to mix. CASTING EMSEMBLE

Apogee has upped the ante yet again, with a definitive upgrade from more affordable units such as the Duet and Quartet. The Ensemble incorporates an ESS Sabre32 32-bit Hyperstream DAC with “Time Domain Jitter Eliminator”. What’s all that mean? It’s a new generation of audio digitising, that’s what! And while the Ensemble doesn’t quite outshine Apogee’s flagship Symphony I/O, it comes extremely close in specification. The Ensemble pulls a THD+N spec of -114dB (at 96k) and a dynamic range of 123dB for its D/A, while the Symphony I/O manages -117dB and a dynamic range of 129dB. There’s really not a lot in it. The resulting sonics are something to behold, with Apogee’s usual smooth and crisp high frequency reproduction, but with a bottom end that’s simply glorious. You’ve never heard your kicks captured and replayed like this before. Well, certainly not at this price. All in all, you’re going to have to look hard to find an interface which offers all the Ensemble can. Not at a price like this. Truly, yet another watershed release for Apogee, and as I mention, to be met wholeheartedly with open arms. There’s one thing which could make the Ensemble Thunderbolt better — a set of eight Apogee preamps ready to patch straight into the unit. Come on Apogee...

“ Spatial detail without loosing the big picture.” Dr. Antti Sakari Saario Head of Music, Falmouth University

“beautifully honest.”

For all demo and sales enquiries, contact: Federal Audio sales@federalaudio.com.au AT 41

REVIEW

KRK Rokit RP8 Generation 3 A new generation of Rokits. We have lift off. Review: Christopher Holder

NEED TO KNOW

As with previous Rokits thereâ&#x20AC;&#x2122;s plenty of balanced/ unbalanced connectivity around the back, along with four-position high and low tone controls.

Price (each) RP5G3: $319; RP6G3: $419; RP8G3: $529 RRP

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Contact AMI: (03) 8696 4600 www.gibsonami.com

Pros Detailed sound stage Plenty of power without distortion

Cons Too much HF sizzle for some

Summary A remarkably competent nearfield monitor at this price point, with all the grunt most studios could wish for. Its character is in the highs and low-mids. Make up your mind if you favour this flavour.

If one manufacturer had totally nailed it. If there was a studio monitor that had the perfect combination of linearity, price, translation qualities, detailed sound stage, then I wouldn’t be writing this review. The truth is, no one has nailed it. And even if you spend $100,000 you won’t have nailed it because someone else’s $100k speakers will offer something just as amazing, but different. The truth is, there’s still room for studio monitor tribalism: Adam versus Dynaudio versus Quested versus ATC versus JBL… and on it goes. And it’s not a case of a Nike versus Adidas where marketing or celebrity endorsements win, there are real differences in tone and performance. Not ‘good versus bad’ differences but flavours. That doesn’t bother me, in fact, I like the fact there is no right or wrong, it’s all about what works for you. When I needed a no-nonsense pair of active monitors after getting my shed back into a condition for electronic music production I couldn’t see past the Event 20/20bas V3s. They’re an eight-inch (well, 7.1-inch to be precise) bi-amped, bass reflex two-way monitor. They don’t look anything remarkable, but they pack a real punch and represent amazing value. So it was with some interest that I put up the KRK Rokit RP8 G3s: an eight-inch, bi-amped bass reflex two-way monitor also on its third version/ generation. How would they stack up against the Events? The Events have been on the market for a couple of years and, as a result, are a bit cheaper than the newer KRKs, but I see them as direct competitors, gunning for a very similar market. After extensive testing, there’s gotta be a winner, right? FATTEST BOTTOM?

KRK built its name in the ’90s by being one of the first to strap an amp to the back of a nearfield monitor. It also immediately captured attention for its use of Kevlar in its woofers (selected for being light yet rigid), and titanium for its tweeters. The result was some beefy studio monitors with big controlled bass and razor sharp highs. And people liked it. The hip hop studio community, especially, adopted Keith R Klawitter (Mr KRK) as its Patron Saint of Big Bottom. KRK first launched its Rokit marque nigh-on 10 years ago now, providing a more affordable introduction to the KRK family. Gone are the titanium tweeters, replaced by one-inch soft domes. The KRK slot-shaped bass port stays, and the curvaceous MDF cabinets are solid and good looking. As with previous Rokits there’s plenty of balanced/unbalanced connectivity around the back, along with four-position high and low tone controls. The G3s also ship with a foam base to the cabinet to go some way to isolating the cab from whatever you’re mounting them on. HEAD TURNER

Spark up the Rokit RP8s and it’s impossible to not be immediately impressed. The RP8s spec out

RP8G2 Review Issue 66 (2009) Launch Price: $749 each (RP8G2) What we said then: While I was still a little unnerved by the projection of their low midrange, I eventually got used to it and began to enjoy the quality of the sound they offered. The stereo image was represented extremely well, and though these are nearfields, they seemed significantly more forgiving to directionality than my current combination. — Michael Carpenter.

at 35Hz to 35kHz ±2dB, which is an enormous frequency range. And with 100W of bi-amped power on offer (for 109dB peak SPL at 1m) there’s plenty of juice. The extended high frequency range (although, obviously not directly audible at 35kHz!) is clearly evident and the voicing adheres to the KRK legacy — there’s nowhere to hide in the upper register. The tweeters and the new HF waveguide ensures the Rokit’s reputation for superior stereo imaging is maintained — it’s a meticulously presented sound stage. Sibilance, cymbal fizz and reverb tails are as clear as bell. The Event 20/20bas V3s are renowned for their lightning quick response, which translates into more impact from a kick drum beater or the toms, while baritone male vocalists are a little more forward. The KRKs still have the welly in the LF but not necessarily quite the same impact. I’m talking about degrees here, not night and day. And this isn’t due to larger cabinet volume (they’re almost identical). Certainly, at 17kg versus 12kg, the Events carry more cabinet weight (presumably through MDF bracing, as the Class A/B amp sections don’t represent much of a weight impost), and throw that weight around responsibly. On the other hand, the KRKs high-mid emphasis means female vocals are slightly more forward, as are the plucked notes of an acoustic guitar, for example. The Event’s higher register on the other hand are well handled, with plenty of detail but not ‘laser’ detail. Out of the box the Events are louder than the KRKs. Checking the specs, sure enough, the 20/20bas V3 have quite a bit more amplification behind them with 80W units on each driver, while the Rokit8 G3s have a 25W amp on the HF unit and a 75W amp driving the lows. Matching the levels of the Rokits meant pushing the faders on my console by about 8dB. Naturally, my initial impression was: ah, my trusty Events clearly have more guts. But messing about with gain, I can confirm what the specs suggest, both speakers are

RP6 Review Issue 50 (2006) Launch Price: $999 a pair (RP6) What we said then: While a lack of low frequency information was my main gripe, they delivered a very well-articulated and ‘biting’ mid-range which I enjoyed working with, and also a very smooth and detailed top end which did not become ear-frying during long mix sessions. The Rokits would no doubt shine on rock material of the punk and thrash variety, or, with the addition of a subwoofer, on just about anything else. — Greg Walker.

about as loud as each other when you push the throttle. In fact, if anything, I’d suggest the Events begin to lose their composure before the KRKs when pushed super-hard. WHERE IN?

As you’ve no doubt observed, I’m a fan of my 20/20bas V3s. I like their big bottom and refinement. For the price I was very sceptical that another nearfield monitor could hold a proverbial candle to them. And in my initial testing all my suspicions about the KRKs seemed to be realised. Too much sizzle and hi-mid action that certainly turned heads but ultimately lead to fatigue. But I soon realised my criticisms and preconceptions were as much about getting my review criteria ducks in a row. I had the Events on my oh-so-solid Foundation speaker stands and the KRKs on the meter bridge. Then I swapped. And with each swap I’d screw my face up wondering how to adequately read the tea leaves. It wasn’t until I put one of each monitor on their own stand that the real story became clear. Both of the nearfield monitors are class acts and for certain mixes they were almost interchangeable. Ultimately I settled on backing off the Rokits’ HF tone control for my tastes. But the exercise reminded me how the sound of these (or any) monitors is as much about positioning and the room as it is about ‘how good’ the loudspeaker is. Bear this in mind if you’re conducting your own in-store demo. If you fall in love with a pair of monitors, then be prepared to spend the time bedding them down into your room. I also reflected on just how good both sets of monitors are for the price. Like I mentioned, the Rokits are a little newer in the market and a bit more expensive as a consequence. That will swing some decisions. But if you do choose the Rokit RP8 G3s, you won’t be sorry. You’ll join a proud cadre of KRK aficionados, and I can assure you: that’s not a bad place to be. AT 43

REVIEW

PHOENIX AUDIO NICERIZER JUNIOR Phoenix Audio’s new streamlined take on the summing box concept is light on features but big on sound. With just two outputs, 16 buttons and pan pots, the Nicerizer Junior puts Greg Walker’s preconceptions to the test. Review: Greg Walker

NEED TO KNOW

Before I get into the specifics of the Nicerizer Junior I’d like to clearly state up front that I’m a longstanding ‘Summing Sceptic’. I’ve never really bought into the idea of dumbing down the classic analogue mixing console signal path to the point where all the fun stuff is gone (faders, equalisers, compressors, routing matrices, auxiliaries or buses anyone?). I’ve always felt that what’s left is a dreary box that doesn’t amount to much more than an adding machine with a couple of transformers in it. I know we’re all trying to cut costs but is this really necessary or desirable? It might sound okay, but is it worth shelling out serious money for? You can imagine my disquiet then when the box that arrived from Mixmasters for my perusal turned out to devote 16 of its 18 red aluminium knobs to

PRICE $2772 CONTACT Mixmasters: (08) 8278 8506 or sales@mixmasters.com.au

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PROS Smooth dynamics & introduces very useable harmonic saturation. +8dB input gain switches add extra oomph. Extra stereo outputs for parallel treatments.

panning! Yep, the Nicerizer Junior is a 2RU box that can put a source into the right speaker, the left speaker or even somewhere inbetween! Okay, I’m being facetious here, but as I wired the thing up to my rig I kept thinking to myself ‘this better do something special or I’m going to take the whole summing thing down next issue!’ SUMTHING FOR NOTHING

Where I had to give an initial tick to Phoenix Audio was in the connectivity department. The two eight-channel D-sub input connectors on the back of the Nicerizer Junior are a no-brainer for a quick analogue hook-up. Everyone who makes multichannel audio gear should use this format and then we can all buy D-sub patch bays and the world will be a saner,

CONS No inserts. Pan controls are redundant in many mix situations. No transformers on the 16 inputs.

less cluttered place. I connected eight channels from my UA Apollo and another eight from my RME converter into the Nicerizer’s inputs, then ran the XLR stereo outputs from the Nicerizer through a pair of tasty JLM PEQ500s on their way back to the Apollo’s inputs for some analogue EQ sweetening. The Nicerizer Junior has an additional set of XLR stereo outs for parallel processing and a master volume pot for each of these stereo pairs. Here I immediately wished there was a master bus insert so I could strap my EQs across both pairs of outputs prior to hitting something in the nature of heavy outboard compression on the second output. I also wanted inserts on another set of D-subs (even eight channels would have been handy) in order to integrate more of my outboard nick-

SUMMARY The Nicerizer’s small feature set is redeemed by great-sounding harmonics and a relatively affordable price tag. The variable gain setting switches allow for some play in the gain structure and there are 16 pan pots to play with. A one trick pony, but it’s a pretty good trick.

nacks without having to wire them inline, but of course Phoenix Audio has eschewed those kinds of options with this ‘Junior’ model in its summing line and have thus kept costs down. Phoenix Audio’s original version of its Nicerizer 16 (now out of production) had transformerbalanced inputs on every channel as well as the outputs. It’s worth noting that the Nicerizer Junior evolved from version two of the Nicerizer 16, and incorporates the same Class A transformerless input stage. Where the two units differ is in the ‘16’ model’s provision of features such as master bus inserts, metering, XLR inputs, headphone monitoring and stereo width control. Of course, all this comes at a price and Phoenix Audio has obviously seen a gap in the summing market for something more stripped back and affordable. Given all this, I decided to keep it simple myself and avoid compression in the audio chain so that I could really hear what the Nicerizer Junior offered in terms of sonic qualities. Panning and master volume aside, the only other features on the Nicerizer Junior’s front panel apart from the red backlit power switch are the 16 white +8dB input gain boost switches. When engaged, these boost signal to the circuitry and help induce higher levels of harmonic colouration in your chain. SUM & DIFFERENCE

Once I started playing around with the Nicerizer Junior on a variety of rock, pop and electronic material I had to admit to myself that it was sounding pretty good. When I A/B’d various Nicerized and straight digital versions of the same mixes it was quite clear that the outboard box sounded, umm…nicer. At the standard gain setting there’s a subtle but definite sweetening of the mix elements; vocals soar satisfyingly, electric guitars gain a little weight and definition, drums sound a little punchier and more robust and there’s an attractive overall tonal signature and sense of unity about mixes that have gone through the summing circuitry. There’s also a gentle smoothing of transients which is quite pleasing and helps differentiate the overall effect from the straight digital mix. Once the +8dB gain switches are engaged the game changes a bit and the effect of the circuitry becomes much more obvious. Harmonic distortion starts to kick in and there are some great ‘hot’ mix tones to be had, particularly on more aggressive material. A big surprise for me was how much the Nicerizer Junior could compress the dynamic range of the mix. At the more heavy handed end of the spectrum I reckon I was getting around 4-6dB of dynamic range squeezed out of my mix if not more, and the overall effect did indeed have something of the ‘analogue console’ sonic quality we all lust after. The qualifying statement here though (just like on a Neve or SSL console) is that you have to be careful not to overcook it. As the Nicerizer Jr runs out of headroom, mixes can start to sound very stressed and transient details get well and truly smeared. By grouping all the channels of the mix in my DAW I could ride the whole thing up and down until I found the sweet spot in terms of clarity,

saturated harmonics and transient response. Unlike on a console (where you can drive individual sources at different levels using input gain and channel faders) you don’t really have the option to selectively cook some of your sources more robustly using the +8dB setting. All the heavy lifting is done by the transformers at the output stage, therefore almost all the harmonic ‘drive’ on tap happens downstream from the inputs and after the summing process itself. On one of the singles I was mixing during the review period, I found the +8dB setting was driving the drums too hard, squashing the transients down too much where I wanted to keep them open and dynamic, so I had to run the whole mix at a lower input gain setting to keep the particular push-pull of that mix working the way I wanted it to. I can imagine some people would find the panning controls useful but I have to say that once I had them panned hard left and right to accommodate the stereo pairs coming out of my DAW I never touched them again. Even when mixing with mono lead vocals and bass I always run these sources and their associated effects (inevitably in stereo) through a stereo pair. Personally I find that most contemporary mixes allow no room in a 16-channel setup for luxuries like mono kick and snare (not when you’ve got five pairs of guitars, three stereo keyboards, tons of backing vocals, electronic samples, percussion,

and effects, several different treatments for drums and lead vocals in different parts of the song). I would happily have individual panning for just four channels and sacrificed the rest for that lovely width control on the Nicerizer 16 v2. This minor gripe aside I had a lot of fun with the sound this summing box delivered and the fact it got used on two mission-critical single mixes speaks for its professional quality and ‘vibe’ factor. PLAY NICE?

While many would claim the digital versus analogue debate has been well and truly won by digital, to my ears there are undoubtedly some things analogue still does better. Top of that list would be blending sounds together and endowing them with a pleasing euphony. For that reason the Phoenix Audio Nicerizer Junior did ultimately convince me of its value in the studio. While it took a while for me to warm to its charms, I now shudder to think of it disappearing back into the courier’s van. It barely interferes with the digital workflow at all and serves up quality sonic results. It’s not the magic-bullet ninja box all on its own and it is frustratingly lacking in features (such as channel and master inserts) but in the fight against harsh digital mixes the Nicerizer Junior is a real weapon that delivers a decent lick of the big console sound at a fraction of the cost. AT 45

REVIEW

Yamaha DBR/DXR Series Powered Speakers Nexoâ&#x20AC;&#x2122;s voice control gives Yamahaâ&#x20AC;&#x2122;s plastic boxes plenty of range.

NEED TO KNOW

Review: Mark Woods

Price DBR: $699 (10), $799 (12), $899 (15) DXR: $1099 (8), $1199 (10), $1299 (12), $1399 (15) DXS Sub: $1299 (12), $1399 (15)

AT 46

Contact Yamaha Music Australia: (03) 9693 5111 or au.yamaha.com

Pros Compact & lightweight Powerful at over 1kW Effective DSP

Cons No top handles Horn throw a bit slim for true monitor use .

Summary The DBR and DXR range of Yamaha powered speakers bring some Nexo punch to the entry and mid-level prosumer market. Both are almost identically powerful, flexible and capable speakers, with the DXR delivering smoother extension at both ends of the spectrum.

Yamaha now owns a large part of French speaker manufacturer Nexo and while it may be a little shy about details, it’s happy to mention the influence of Nexo’s engineers. Specifically, how their involvement led to some new approaches in the development of the drivers, cabinets and DSP for Yamaha’s new speakers. The DBR and DXR range of powered speakers replace Yamaha’s ageing MSR400 boxes and allow it to compete in the current, hotly-contested, prosumer market. They come in different sizes, with woofers from eight (in the DXR) or 10 (in the DBR) to 15 inches, plus 12- and 15-inch subs in the DXR range. I got to try the 12-inch version of each range. Both the DBR12 and the DXR12 are twoway ported designs utilising Yamaha’s FIR-X linear phase crossover, and they boast impressive, almost identical, technical specs. I’ll accept at face value the claimed 1dB maximum SPL difference between them (131dB vs 132dB) as I don’t want to find out what they sound like at that volume. A DBR OF DIFFERENCE

The Indonesian-made DBR12 is the entry level model and it’s available for $799. Yes, there are cheaper speakers available, but they’re likely to be lower quality and for any sort of demanding application they’ll probably only ever sound fair. Once you get to the DBR level you’re entering the world of professional sound quality and you can expect decent components, power and reliability. In keeping with their entry-level status the DBR range comes in polypropylene enclosures… so they look like plastic boxes. They are not unattractive though — the shape is functional, and the front features a distinctive steel grille and prominent ‘three tuning forks’ logo. Comfortable handles with useful finger grips are set into the sides, while the rear is shaved across the back corners to point the speaker up at a 50-degree angle if they’re being used as floor monitors. The recessed rear panel contains the user controls

and connections and they conform to the current prosumer standard of several inputs that can be mixed together, plus a handful of EQ options. The DBR12 has two input channels: Channel 1 has a combo balanced XLR/¼-inch jack socket and switches from mic to line input level; Channel 2 has a combo input plus two RCA sockets. These can be used at the same time, with a synth in the combo line input and a laptop in the RCA inputs for instance. The only compromise with this is the lack of separate volume controls so their respective levels would need to be controlled at the source. The single XLR output can be switched to send Channel 1 straight through (to create a stereo system) or send a mix of both channels to another speaker. Sound-shaping options include a two-stage hi-pass filter and DSP settings that can be switched to flat, FOH (added lows and highs) or Monitor (cut lows, added hi-mids and highs). I liked the way the connections and controls are spaciously laid out and easy to find — it’s often dark back there, and there are dangers in groping around… beware the sudden-death switch (mic/line selector) on Channel 1. I’d like to see as standard a little switchable light above the rear panel. MARKING UP TO DXR

The Chinese-made DXR12 is the mid-level model, and at $1299, is right amongst the leading brands. This category appeals to buyers that expect high audio quality, class-leading features and solid performance in a range of applications. In keeping with this increased expectation the DXR range comes in a sharp looking ABS enclosure. It’s pretty much the same shape and dimensions as the DBR range but the sharper edges and smooth finish give it a more upmarket attitude. The wrap-around steel grille and white power light help too. Under the cabinet there’s the luxury of a second polemount option, pointing down seven degrees. The rear panel is similar to the DBR12, with the same EQ options, but offers an extra input so you get

the choice of XLR, ¼-inch jack or RCA, each with separate volume controls. There’s a recognition that these types of speakers will often be used without a separate mixer, and it’s great for small/solo acts to be able to mix themselves from the back of the speaker, but no phantom power on the vocal channel may be limiting. DIALLING UP VOICE COILS

In use, both models have similar voicing with the main difference being the quality of the delivery. The DBR12 is a willing performer with a strong mid-range presence and more than enough power for the drivers. Vocals are clear at all levels with the onboard D-Contour multi-band compressor earning its money dealing with the dynamics of live vocals. You can hear the compressor working sometimes but it effectively stops the mids from distorting on vocal peaks and makes for a smoother listening experience. The LF driver has a twoinch voice coil and produces enough low-end for music playback, especially with the FOH EQ setting selected. At high levels the low end gets overwhelmed before the mids but the processor stops it from getting too untidy. Horn dispersion is 90 x 60 degrees and while the one-inch throat HF compression driver is a little grainy above 6kHz, the horn flare does a good job of spreading the high frequencies evenly across the front of the cabinet. The DXR12 has an X (for Nexo?) in its title and sounds better for it. It’s noticeably cleaner than the DBR12 at all levels and the graininess of the DBR12’s horn is replaced by a smoother, extended response in the DXR. Its claimed maximum volume may be only 1dB higher than the DBR12 but because it stays tighter at high levels it’s effectively a greater difference. The LF driver has a bigger 2.5inch voice coil that delivers a richer low-end with less distortion at high levels. Does it sound like a Nexo? Not really, but that doesn’t mean it’s not a good speaker in its own right, and doesn’t mean it hasn’t benefitted from a little Nexo fairy dust. AT 47

Both can be pole-mounted or flown via M10 eyebolts.

The XLR output can be switched to send Channel 1 straight through (to create a stereo system) or send a mix of both channels to another speaker (to replicate the mix in both speakers).

Though not strictly an approved feature. Channel 2 combo and RCA inputs can be used at the same time, with a synth in the combo line input and a laptop in the RCA inputs for instance. The only compromise with this is the lack of separate volume controls so their respective levels would need to be controlled at the source.

Power comes from Class D amps, 1kW for the DBR12, 1.1kW for the DXR, and there’s lots of 24-bit/48k DSP behind the scenes.

OUT ON THE CIRCUIT

It’s festival season for me so I got to try these speakers in a variety of applications. They travel easily, and they’re light; the DBR12 is less than 16kg, the DXR12 just over 19kg. They’re easy to grab by the generous side handles but there’s no handle on the top of the cabinets, not a huge issue but I use them often when fitted. Speaker covers/ travel bags are almost essential for this class of speaker — I don’t think many of them travel in the back of trucks, rather they either get used in installations or travel to the gig with the band. They need bags or they get scuffed up in no time. The optional covers for these are padded for protection and have a front mesh so the speaker can be used with the cover on. As small FOH speakers they both worked easily, with the quality difference between the different models only really noticeable on either loud or very detailed sources. Both can fill a small room — the AT 48

amount of bass is the limiting factor but they’re surprisingly good for a 12-inch woofer and if you need big bass then you need a DXS sub, or at least the models with a 15-inch LF driver. As delay speakers for a bigger system at the new Blackwood Festival they had a good, even dispersion pattern and ran happily without any external EQ. Clarity and intelligibility across the vocal range is their strongest feature and that also applies when using them as monitors. They’re a great size for sidefills on small-medium stages, and there are 50-degree angles on both rear edges so a pair could be placed in a mirror-image pattern with the horns on either the inside or the outside. The Monitor EQ setting helps in controlling the excess low-end you don’t need or want on stage, but the fixed dispersion of the horn is a compromise. On a stage, the 90-degree horizontal throw of the horn becomes 90 degrees of vertical throw, with only part of it aimed at the singer. Plus

there’s the problem of entering and leaving the horns relatively narrow horizontal angle when moving around the stage. The answer is a rotatable horn, then they could they really claim to be equally suitable for floor monitors, until then… they’ll do in a pinch. At the Maldon Folk Festival this year the Yamahas started off on the floor but they really started to work when I switched them to sidefills — the even coverage and high resistance to feedback winning favourable comments from the acts. The DBR and DXR series of powered speakers are built to a price but they’re well designed, well built and equipped with all the controls and protections expected in their classes. Yamaha has a reputation for making bullet-proof gear and these come with a five-year manufacturer’s warranty. The DBR series will sell on price and shouldn’t disappoint but if you can hear the difference clearly, then the DXR series is worth the extra expense.

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