AudioTechnology App Issue 27

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

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

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

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

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

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All material in this magazine is copyright Š 2015 Alchemedia Publishing Pty Ltd. Apart from any fair dealing permitted under the Copyright Act, no part may be reproduced by any process with out written permission. The publishers believe all information supplied in this magazine to be correct at the time of publication. They are not in a position to make a guarantee to this effect and accept no liability in the event of any information proving inaccurate. After investigation and to the best of our knowledge and belief, prices, addresses and phone numbers were up to date at the time of publication. It is not possible for the publishers to ensure that advertisements appearing in this publication comply with the Trade Practices Act, 1974. The responsibility is on the person, company or advertising agency submitting or directing the advertisement for publication. The publishers cannot be held responsible for any errors or omissions, although every endeavour has been made to ensure complete accuracy. 10/02/2016.

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COVER STORY

Get Ready For Summer: Howard Page’s Ultimate PA Tuning Guide

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ISSUE 27 CONTENTS

42

Mixing Hermitude’s Dark Night Sweet Light

Mix Masters: Downtown

48

Neil Young & the Ultimate ‘DIY’ Large-format Console

Reggae Meets Rhumba: Jamaican & 34 Cuban Legends Together for the First Time

Extreme Isolation Ex-29 Headphones AT 6

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Zoom F8 Multi-track Field 56 Recorder

Bose F1 PA System

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GENERAL NEWS

APOGEE SYMPHONY I/O MKII INTERFACE Apogee has introduced a second generation ‘Mk II’ version of its flagship audio interface. Symphony I/O Mk II is a multi-channel audio interface with Apogee’s latest AD/DA conversion, modular I/O (up to 32 inputs and outputs), intuitive touchscreen display and optional microphone preamps. The new Symphony I/O Mk II lets you choose direct connectivity to one of three different platforms — Thunderbolt, Pro Tools HD or Waves SoundGrid network. Latency specs are impressive — 1.35ms with Thunderbolt and Logic Pro X. Apogee says

Symphony I/O Mk II raises the bar on all of the company’s flagship products that have come before it, offering even better audio clarity and sonic transparency. When equipped with either the brand new 8×8 or 16×16 Mk II I/O modules, Symphony I/O Mk II positions itself among the best-performing multi-channel audio interfaces available for Pro Tools HD or Thunderbolt based systems.

Sound Distribution: (02) 8007 3327 or www.sounddistribution.com.au

ASTON HALO REFLECTION FILTER UK-based Aston Microphones is making a splash in the audio world that started with the release of the Origin and Spirit condenser microphones. Now, the company has released the Halo, introduced at NAMM — a product that boasts the latest in reflection filter technology. The patented design is supposed to absorb frequencies much more linearly than its competition, while the dome-like construction catches more sound than a single

curved panel. The Halo features filtering on both horizontal and vertical planes, and a surface area almost 40% bigger than sE’s popular counterpart.

It’s also designed to be lightweight, so it won’t topple your mic stand. Comes with proprietary ‘easy-mount’ hardware.

Link Audio: (03) 8373 4817 or info@linkaudio.com.au

AUDIO-TECHNICA: NEW MICS AND IEMs Audio-Technica introduced a number of new microphones at NAMM. The Artist Elite AE2300 is a dynamic cardioid instrument mic with AudioTechnica’s double-dome diaphragm design.

Equipped with a switchable low-pass filter for getting rid of high-frequency noise, the AE2300 has a low profile design and can handle high SPLs. The ATM230 is also a dynamic instrument microphone, ideal for use with high-SPL percussion instruments like toms and snare. The hypercardioid

polar pattern is good for rejecting unwanted spill in drum kit recordings, or on the stage for live sound applications. Also new from Audio-Technica is the the addition of three professional in-ear monitors, the ATH-E70, ATH-E50 and ATH-E40. In the sonic vein of the M-Series, the headphones are compatible with Audio-Technica’s M3 IEM wireless systems. Availability is from April 2016.

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

KORG MINILOGUE Korg has released a new polyphonic analogue synthesizer called the Minilogue. The sleek 37key slim-key synth’s analogue circuitry has been designed from the ground up with unique wave shape capability allowing you to fine tune the oscillator’s harmonics. Various modulation types are also available including cross modulation, oscillatory sync and ring mod. A polyphonic step and motion sequencer is a neat addition, along with the on-board tape-style delay, multiple sound shaping and filter options, and even an OLED oscilloscope display. Eight separate Voice

Modes allow the Minilogue’s four voices to be flexibly configured in different ways. You can save your settings in 200 program memories, with the Minilogue shipping with 100 preset sounds onboard. Design features include an aluminium top panel, chassis-mounted pots, rubber-coated knobs, and a real wood back panel. The Minilogue will feel at home both in the studio and on stage with MIDI, USB MIDI and Audio Sync connectivity available, including direct sync.

CMI Music & Audio: (03) 9315 2244 or www.cmi.com.au AT 9

LIVE NEWS

CLAIR GLOBAL COHESION SERIES Clair Global announces its new proprietary flagship system: The Cohesion Series, designed for both sound quality and efficiency. The system consists of five speaker enclosures that provide high output reference-quality sound within a smaller footprint in both truck space and in the air. “Based on feedback from our clients, we knew that our job was to design a system that does more with less.

Now we offer a 270° coverage system that fits into 30ft of a truck. The system is engineered for speed and safety,” said Shaun Clair, VP of Sales for Clair

Global. The Cohesion Series was also conceived to be a very ‘green’ system. All Cohesion amplifiers, including the CP-118 and CP-218 self-powered subwoofers, are Power Factor Corrected for extremely high-energy efficient operation, both of which allow tours to reduce their carbon footprint significantly. Expect to see The Cohesion at shows with artists such as Carrie Underwood, Billy Joel, Bruce Springsteen, The Weeknd and Black Sabbath.

Clair Global: www.clairglobal.com

HK AUDIO LUCAS NANO 608i HK Audio’s LUCAS NANO family of portable crossover PAs is expanding with the LUCAS NANO 608i, the first high-end, all-in-one system with an iPad-enabled wireless mixer and Bluetooth technology built in. The eight-channel digital mixer integrated into LUCAS NANO 608i’s subwoofer can be controlled in two ways: either directly on the 608i itself using conventional knobs, or wirelessly via a free dedicated iPad app. Signals are easily shaped with features like ‘drawable’ EQs, compressors and high-quality reverb programs.

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Adjustments can be made in the more sophisticated ‘Expert Mode’ or in the intuitive ‘Easy Mode’, and then stored. Weighing just 16.3kg, LUCAS NANO 608i’s discreet profile couples with full range sound to make it a great PA for a variety of stages. The system’s flexibility means it can be deployed as a slim mono column, a classic stereo setup or as a twin stereo system. Ideal for solo musicians, small groups and entertainers.

CMI Music & Audio: (03) 9315 2244 or www.cmi.com.au

JBL VTX M SERIES Harman unveiled a new line of stage monitoring products, the JBL VTX M Series. Designed with a low profile, the M Series utilises JBL’s Neodymium Differential Drive woofers and D2 Dual Diaphragm high frequency drivers, along with a new high frequency Image Control Waveguide for better control and gain before feedback. Two models make up the new VTX M Series line — the VTX M20 and VTX M22. The M20 features dual 10-inch woofers and a three-inch D2 compression driver, while the VTX M22 has dual 12-inch woofers in addition to the three-inch D2 compression driver. The M Series

high frequency section utilises the proven D2430 Dual three-inch compression driver, also found in JBL’s large format VTX V25-II Line Array system.

The woofers in the VTX M Series utilise JBL’s patented Differential Drive technology, allowing for excellent heat dissipation, low distortion and very high output, while maintaining unusually low weight. The speakers are powered exclusively by Crown iTech HD DSP-based amplifiers.

Jands: (02) 9582 0909 or www.jands.com.au

MACKIE DC16 CONTROL SURFACE Mackie has launched the DC16 control surface for its DL32R digital mixer. The combination of the DL32R digital mixer and new DC16 control surface delivers a scalable system with all the benefits of wireless mixing designed from the ground up as an integrated system. With 32 remote-controllable Onyx+ mic preamps and 16 outputs paired with massive built-in DSP, the system is ideal for medium and large channel count applications.

The system relies on Dante for communication between the DL32R mixer and DC16 control

surface, enabling additional networking capability for professional applications. Visual feedback is provided by full-colour channel displays and the ability to dock up to three iPads in the SmartBridge.

Users can grab an iPad from the DC16 and head out to tune the room or dial in monitors and the Master Fader control app knows to switch it to the view that makes the most sense for the user.

CMI Music & Audio: (03) 9315 2244 or www.cmi.com.au

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SOFTWARE NEWS

CEDAR AUDIO FNR MODULE CEDAR Audio recently announced its latest addition to its boundary-defying noise reduction software lineup — the FNR module for CEDAR Cambridge. Based on a new noise reduction algorithm, it is designed to clean up speech recordings suffering from very poor signal/noise ratios, making it ideal for audio forensic and security clients, not to mention post and broadcast.

“FNR is created to be adaptive, reacting quickly to changes in the background noise in café-type recordings, and it doesn’t have anything like as much effect on the wanted voices as previous

filtering systems,” says Alan French, head of CEDAR’s forensic and security division. “I have been testing FNR for more than six months prior to its release, and it has yielded excellent results on genuine casework. It also works very well on recordings made in moving vehicles because, as in the café scenario, the background noises (primarily, the engine and road noise) are constantly changing, and FNR tracks these changes better than anything else I’ve yet heard.”

CDA Pro Audio: (02) 9330 1750 or www.cda-proaudio.com

LURSSEN MASTERING CONSOLE IK Multimedia announces Lurssen Mastering Console, a new mastering software tool developed in collaboration with mastering engineer Gavin Lurssen, Reuben Cohen and the team at Lurssen Mastering in Los Angeles, CA. It’s designed to be an effortless mastering tool for musicians and producers for when a trip to Lurssen Mastering just isn’t in the budget. Lurssen Mastering Console comes with 20 unique mastering templates called ‘Styles’ that have been crafted by Gavin and the team of engineers. Controls include an Input Drive AT 12

knob, five band EQ, and special ‘Push’ knob which lets users ‘ride’ the flow of a song. Adjustments to the Input Gain and Push knob can be recorded and automated. The software’s 88.2/96kHz DSP processing allows for a great degree of sonic detail.

You can export in all popular file formats from AAC to WAV, FLAC and more. It also allows for quick export to File Sharing, SoundCloud and more.

Sound & Music: (03) 9555 8081 or www.sound-music.com

AVID PRO TOOLS 12.5 Avid released three software updates at NAMM: Pro Tools 12.5, a 2.1 firmware update for Avid’s S6 control surface and Sibelius 8.1. For the Pro Tools users out there, the 12.5 update continues the stream of Tools updates promised to subscribers.

It’s the fifth in the last year. 12.5 looked to make good on the central promise of the Avid Everywhere scheme, namely to make cloud collaboration ubiquitous. Not so fast though, it’s only for select

customers. Those lucky ducks on the ‘Early Access’ beta program will be able to share live projects from anywhere in the world. Too bad if your colleague isn’t on the beta program though. For those not on the beta, Avid has announced that the walls have come down on its Artist Community, which is now available to everyone.

Avid: (02) 9931 6841 or www.avid.com

GREG WELLS MIXCENTRIC PLUG-IN Greg Wells’ collaboration with Waves continues with the newly released MixCentric plug-in. The Grammy-nominated producer, songwriter and mix engineer has already put out a vocal plug-in called VoiceCentric, and more recently a plug-in for processing pianos called PianoCentric. MixCentric is designed to be a ‘finishing touch’ plug-in you place on your mix bus. It features EQ, compression and harmonic distortion as a quick solution for enhancing and balancing the tonality of a mix. It’s not a loudness plug-in, and Waves states it should

be used with an instance of a limiter plug-in placed after it to bring up the overall level of a mix in its final stages. Greg Wells says, “There is an enormous amount going on behind the deceptive simplicity of the control knob, including many different settings that come in and out and change as you move the knob position. Play with it and discover the incredible power of this user-friendly plugin.”

Sound & Music: (03) 9555 8081 or www.sound-music.com

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REGULARS

Watchout!

Following protocol to track the latest in audio networks.

Column: Paul Doornbusch

Dr Paul Doornbusch is the Associate Dean, Audio Production Program Leader, at Australian College of the Arts. Paul loves computers so much, he reconstructed and documented the music played

by Australia’s first computer (CSIRAC) while he was a composer-in-residence at the Computer Science department of the University of Melbourne.

audio, we need reliable data delivery or else we get highly undesirable clicks.

There are a number of systems for running digital audio over a computer network. CobraNet and EtherSound are older solutions; Q-Lan, Dante and Livewire are more recent developments; while AVB and AES67 are the latest. Let’s look at three of the main players.

DANTE

I’m waiting for the day some Elon Musk-type figures out how to make wireless recording commonplace. The best we’ve got so far is Mikme — a product of the little Indiegogo campaign that could — and bandwidth is still a limiting factor that hasn’t been solved. For the immediate future, audio will be distributed over computer network cables and equipment.

It’s a natural development. The data is already in digital form and networks and computers are rapidly increasing in performance. Bandwidth isn’t a problem for ethernet like it is wireless. With gigabit ports on every new computer, transporting 400+ channels of 24-bit/48k audio over a single network cable is not unreasonable!

Computer networks work by breaking data into ‘packets’ and then using an intricate set of protocols to get the data packets where they need to go. There are protocols for many purposes (see sidebar), from getting and sending email, to requesting webpages, to streaming video. A little delay here and there doesn’t bother data like emails or webpages. But for So in the future we may well see audio networks like this:

The world did not wait around for manufacturers to build an interoperable system, Audinate’s proprietary Dante audio networking solution basically took over the live sound market because it works well and is available now. Several manufacturers announced support for Dante in their live sound consoles from 2012.

The benefit of Dante is that it works over current networking equipment, particularly network switches, because Dante works at Layer 3 of the protocol stack (see the accompanying sidebar).

Dante offers 512 channels in and 512 channels out of a device, all at 24-bit/48k, over standard network cable (CAT5e). The maximum sample rate is 192k and this reduces the channel count to 128 channels, still not too shabby!

However, as Dante operates over standard network equipment it does not guarantee delivery of packets. Latency is 5ms and can be aligned across devices for accurate synchronisation, but this requires careful network management. With multiple connections, Dante offers click-free

failover to another cable in the event of a fault.

There is also no support for video data in Dante.

Dante, being the oldest of these three, has the greatest support among manufacturers. The long list of names includes Yamaha, Midas, Allen & Heath, Bose, Soundcraft, Lake, Shure, EAW, Avid, AKG, Audio-Technica, Behringer, DiGiCo, Focusrite, Extron, QSC, Presonus, SSL, Studer and more. This makes Dante something of a de facto standard in audio networking, but there are some exciting developments on the horizon. Read on.

AUDIO VIDEO BRIDGING

Audio Video Bridging, or AVB, is an open standard not owned by any single manufacturer. It offers a more plug-and-play solution and overcomes some of the limitations of Dante. AVB operates at Layer 2 of the network stack which has several advantages.

On the downside, it needs special hardware — an AVB-capable switch.

AVB allows for 400 bi-directional channels per device of 24-bit/48k audio data. Significantly, AVB offers guaranteed delivery of AVB packets with only 2ms latency. This is achieved with hardware control reserving up to 75% of the available network data bandwidth exclusively for AVB packets. The downside of this is that you cannot send AVB packets over the internet or between networks because that sort of data is all controlled at Layer 3.

However, as utopian as AVB sounds, equipment has been slow to appear and Dante has increased its

Historical & AES67 Networks: Q-Lan Livewire Ravenna, etc

AES67 Dante Equipment: Yamaha Midas Shure, etc AT 14

Dante

Stanndard Managed Network Switches

AES67

AVB Network Switches

AVB Equipment: MOTU Avid Meyer, etc

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STACK O’ PROTOCOLS

The way computers send information over networks is that each computer has an address, and ‘packets’ of information are sent from one address to another.

It’s a bit like sending a cake to someone in separate slices, one slice at a time.

But instead of being able to whack the whole address on the envelope, you can only put one detail on each. You have to start with their name, then slot that envelope inside another with the street name and number, then into another envelope with the city written on it, and lastly, stuff it all into a final envelope labelled with the state.

little gets lost, or maybe you need all of the cake to arrive in order and none to get lost, but you don’t care if that takes longer. Some network protocols are faster and less reliable (like UDP, often used for streaming media) and some are very reliable (like TCP/IP, used for email and web page transmission) and will resend a packet that gets lost, so that nothing corrupts the message.

The OSI Network Protocol Stack below shows the various logical layers that are needed to push data around a network. Each of these layers typically adds header information (the envelope example). Header information is added as data moves down the stack. At Layer 3 the data is broken into packets, and so on until the data is transmitted over the cable as bits in Layer 1.

Like this: <envelope state: NSW

<envelope city: Sydney

<envelope street name & number: 49 Flinders St. <envelope apartment number: 4 Host Layers

<envelope name: Sarah > >

>

> Once the piece of cake gets to NSW, the post office takes off that envelope and sends it to Sydney, where that envelope is removed and so on until it’s delivered to Sarah. As you can imagine, this adds considerable overhead to the process and it can get complicated if the pieces of the cake should arrive out of order.

In computer networks, each application or piece of network equipment adds an envelope to the packet of data, which is later removed when it has arrived at the next destination. The Open Systems Interconnection model is a concept that standardises and characterises the communication functions for computer communication. You can imagine each of the seven layers adding an envelope with items such as the addresses of the sender and receiver, session for the date, the packet number, the time to live, and — for digital audio data — when it should be played.

Network protocols have evolved different characteristics. You might want the cake to get to the destination as quickly as possible and you don’t care if a

user base in the meantime. Significantly, Audinate has said that Dante will support AVB when it is established. While manufacturer implementations may be slow in appearing, both Presonus and MOTU have recently released AVB equipment, plus Avid and Meyer have had it for a while. Other manufacturers supporting AVB include Apple, Crown, Beyerdynamic, DBX, Soundweb London, Netgear, Cisco, AudioScience and so on.

AES67

The Audio Engineering Society has developed the AES67 standard for audio-over-IP interoperability. AES67 is a Layer 3 protocol that offers interoperability between various competing audio networking systems, such as Dante, Q-Lan, Livewire, RAVENNA and so on. It also identifies common elements with AVB and documents how it interoperates with AVB. With AES67 there is an open standard for audio data over IP. Open standards are good for a number of reasons, and should result in cheaper equipment overall. AES67 allows these competing systems to interoperate, and to get audio data between

Media Layers

Layer Name

Data type Which Layer?

7. Application From the network process to the application

Data

6. Presentation Data representation and encryption

Data

5. Session Inter host communication

Data

4. Transport End-to-end communication and reliability

Segments

3. Network Path determination and IP (logical addressing)

Packets

Dante AES67 Q-LAN by QSC RAVENNA

2. Data link MAC and LLC (physical addressing)

Frames

AVB AES51 CobraNet EtherSound SoundGrid REAC by Roland

1. Physical Media, signal and binary transmission over the cable.

Bits

AES50 A-Net by Aviom vRockNet by Riedel

networks. So if a connection is made between an AVB network or product to an AES67 network or product, then AES67 will allow the audio to be sent to a Dante network or product!

So in the future we may well see audio networks like this: NET EFFECTS

Of course, users drive adoption, and for most Dante already provides a suitable solution. Dante is not perfect and users have reported issues such as clicks and PC firewall issues. It’s a new game of diagnostics with this technology. However, most of the time it works as intended and it is here now. As needs change in the future AVB offers some real advantages and should be easier to manage, use and debug. AES67 allows the interoperability of these systems so there is a way forward regardless of the system you chose (or more likely inherit somehow). The real world may well see several of these systems working together because of a historical mix of equipment.

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QUICK MIX

The

with

Michael ‘Smasha’ Pollard Interview: Neil Gray

Who have you been mixing recently?

Last year, I did FOH for the APIA Good Times Tour featuring Leo Sayer, Richard Clapton, Russell Morris & Joe Camilleri. The tour included a nine-piece band of the country’s best players and weaved around Australia over two months. I attempted to make each show of the original lineup Mondo Rock 33 1/3 Anniversary Tour a hi-fi experience. I also got to mix Daddy Cool following their induction into the hall of fame at The Age Music Awards.

For smaller analogue gigs I travel with go-to processing racks to ensure I have my preferred preset FX and dynamics processing.

Those days are increasingly rare.

Other bands you’ve worked with?

Most memorable gig or career highlight?

My regular client over the last six years has been Ross ‘Eagle Rock’ Wilson. He keeps me pretty busy. It’s a bonus that his band are not only great players with extensive experience but excellent humans.

It makes all the travel and time together so much easier. I’ve been working with Speed Orange, and I’ve mixed a couple of shows for Five Mile Sniper. They’re kind of an indie super band featuring past members of Ice Cream Hands, MotorAce, P76 and Pretty Mess.

Three highlights come to mind. At the turn of the century, I had the opportunity to mix chart topping NZ band The Feelers at the first outdoor rock concert in Hong Kong when it returned to the control of China. It was a crazy show with armed uniform guards around the stage and a drummer from a Japanese punk band getting arrested for playing the set in little more than a sock-jock.

How long have you been doing live sound?

I started out in 1988. While completing a sound course, I threw myself at every gig I could find. In Brisvegas that meant a lot of pubs with Yamaha desks and Lexicon SPX90s in the rack. I then lived and worked in Sydney and London for a few years each before settling in Melbourne. Melbourne has worked out to be a great city to be based in, at least in terms of industry activity and opportunities.

What’s your favourite console?

Currently I’m most comfortable with Avids. There are few if any surprises; the show file always loads, you don’t have to rely on the screen for all information and can mix with a sense of tactility (handy when sun glare strikes at some outdoor events), and they’re easy to source. My long-standing use of Pro Tools plays some part in my familiarity with the plug-ins. The reality is I mix on whatever is supplied. The past month of gigs has had me mixing on products from Midas, Soundcraft, Digico, Allen+Heath, Yamaha and Avid.

Favourite microphone or any other piece of kit?

I never leave home without a bunch of USB showfile sticks, Technics cans and my Audix D6. For me, this mic provides a result every time regardless of system and style and I find it quite versatile; it’s my friend in small and large systems. I’ve also noticed that drummers like the D6 character back through their monitors/IEM. I recently tried out the Telefunken M82 which is a very flexible, great sounding mic for kick and instruments; like an RE-20/SM7 hybrid.

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More recently mixing the Time of My Life superband featuring Daryl Braithwaite, Joe Camilleri, Ross Wilson and James Reyne at the base of Big Red — a 30m-high sand dune on the edge of the desert — was a special experience. Big Red made an impressive stage backdrop, and acoustic properties of sand everywhere was unique. I felt for the production company; they would have had sand in every piece of gear!

Lastly, the opportunity to facilitate the Face The Music Conference Q&A session with Steve Albini. Having admired much of the studio work he has engineered, it was a rare privilege to ask a bunch of questions and chat about his perspective on sound engineering.

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

If mixing in digital, I enjoy refining the production with snapshots/ scopes, save/recall and well-designed plug-ins that allow more mix production detail, such as easily manageable side-chain busses and routing. Certainly most systems now offer excellent detail and coverage that can be further developed within the mix to bring out the character of the artist and each song. It satisfies the studio production engineer in me.

What are three mixing techniques that you regularly employ?

1. Parallel compression for both vocals and drum shells. Yep, we all love it, and is now a widely applied technique. It just works so well in containing but highlighting critical mix elements without flattening the overall dynamics. I’m enjoying those desks and plug-ins that offer the blend control right there at the compressor.

2. Mic choice. I like to spec or travel with the same mics, especially for

BOLT OUT OF THE BLUE Supercharge your desktop DAW system Ultra-low latency Lightning quick 10Gb/s Thunderbolt transfer rate 2 x great-sounding preamps Full metering Auto gain feature Pro-grade converters drums. It translates to a repeatable mix that also compliments the recall features of digital desks.

3. Mix from the vocals back. The vocal is the star; build the mix up with the vocal as the focus and use treatments to sit the BVs around it. Depends on the band but some tasty pitch thickening, panning and Haas-zone delay for width and depth are my go-to building blocks when the desk, DSP and time allow. The Mondo Rock tour really got me investigating how to evolve vocals and lush BVs — it was back to the ’80s!

What have been game changers for you in the last 15 years?

Tough, but I’ll go for offline editors, great system techs and Radial products.

To remotely prep a desk in advance is now a normal and essential routine for most shows, most weeks. The range of digital desk products that are supplied at shows over the year is quite varied.

Each brand is trying to stand out in a crowded marketplace with its own take on the digital mixing desk, which can be a great distraction when the core job is to create a mix. It sometimes feels like the desk interaction is a distraction to the mix workflow, until I get some time on it and the familiarity is there. So to be able to check out the desk in the software editor at home, prep the desk layout, and do essential settings ahead of time is a winner.

Many current speaker systems benefit from the application of prediction software and attention to rigging to get the most of the system in a given venue. A great System Tech may make or break how my mix decisions work out and how that mix interacts and covers the venue.

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And Radial, it makes rock solid solutions to everyday signal capture/ control. There’s always a JDI in my Pelican case, it never fails!

Any words of wisdom for someone starting out?

In the ’90s, I recall a sign stuck above the FOH rack at the Melbourne venue Revolver — “Gain Structure, it’s an Immutable Law, learn it, live it”. Good advice. More so these days for those entering the industry; to survive and respect the digital path. From the DAW in the bedroom to the largest show, the management of signal level is crucial, as is the interpretation of metering.

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FEATURE

Tuning & Optimising Large-Scale Concert Sound Systems Losing something in translation? Learn Howard Pageâ&#x20AC;&#x2122;s signature PA tuning method and you can be confident that every seat in the house will hear your mix as it was intended to sound.

Tutorial: Howard Page

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WHAT DO YOU NEED?

Over the years, I’ve prescribed some low-end diets for AudioTechnology readers struggling to unveil the clarity in their PAs. But even though the tuning of some operators would have you believing differently, there is more to a sound system than just sub bass. During this last Summer European tour with Sting I conducted a number of impromptu masterclasses at FOH with the local sound company’s crew. Many times the participants suggested I put the documents in print, naturally I thought of AudioTechnology. Herein lies the method I’ve developed for reliably and consistently tuning and optimising any large scale concert sound system. According to at least one badly translated Swedish review, the sound on the Sting tour was ‘powerful and stable, strong without doing evil in the ears.’ If you like, you can follow my method and stop doing ‘evil in the ears’.

• Computer with FFT measurement system software such as Smaart or EASE SysTune installed.

• Dual-input computer audio interface.

• Flat response omnidirectional reference microphone.

• Custom interface cables as per the connection diagram below.

SYSTEM PRE-TUNE CHECKS

Before attempting to tune any sound system it’s vital to very carefully check all the various sections of the system; ensure all the rigging, wiring, patching and powering up have been performed correctly. Then listen to each section of the system in isolation to verify the complete integrity of the entire system. WHEN SETTING UP IN A NEW VENUE YOU SHOULD NEVER ASSUME EVERYTHING IN A LARGE-SCALE SOUND SYSTEM WILL WORK PERFECTLY WHEN FIRST

THE CONCEPT

TURNED ON!

My method here removes many of the variables that have made previous system tuning approaches extremely unreliable and caused them to deliver inconsistent results from venue to venue and system to system. Some of these variables are personal listening tastes and tuning methods, system brand and design, differing venue styles and each venue’s acoustic properties.

By removing as many variables as possible in the signal chain after the mixing console all the creative focus is now placed where it belongs — at the actual mixing console. An important side benefit of this method is that any recordings, video feeds, and live broadcast mixes fed directly from the mixing console are far more accurate and more closely reflect the actual live sound of the show.

The goal of this method is to have the sound system reproduce as closely as possible the same tonal balance and perspective — ie. ‘the mix’ — generated by the operator at the mixing console’s output.

You must be extremely careful at this first system turn on/check out stage to ensure you’re not starting to tune a system that is not working correctly. There are so many small elements that can be overlooked. If the system is being run with digital AES wiring it’s essential to check that all processors are seeing and locked to a digital AES source. Do this every time the system is set up as having any portion of the system automatically swap over to its analogue input will create some very strange sounding artifacts indeed! A favourite board microphone plugged into the console with some, ‘Check one, twos’ while panning between the left and right stereo outputs will tell you very quickly if the system, in its raw state, is at least reasonably stereo matched on the Mains.

THE BASIC PRINCIPLES

Most concert sound systems used in large venues consist of a larger main system with multiple separate support systems that must be tuned

and optimised to work together as a complete system. As an example, a typical arena system will comprise a large main left and right stereo front system (the Mains); a set of dedicated subs systems; in some cases an additional pair of side hangs to increase the horizontal coverage beyond the limits of the main system; a dedicated front fill system; and in some cases, a separate set of delay systems to reinforce the sound for audience members situated beyond the real limits of the main system’s coverage area.

FOH sound engineers primarily use the Mains of the system as the reference to create the show’s final mix — the same mix the audience across the coverage area of the full system will hear. It is therefore vital that all the other support systems absolutely relate in both level and tonality to what the sound engineer is hearing from the Mains.

THE ‘CREEP’ FACTOR

A frequently overlooked danger when tuning and optimising the same sound system in a touring situation is the ‘creep’ factor. This very subtle but serious problem occurs when typically, at the beginning of a tour, the system is first tuned in a venue for a particular setup of the system. This could be all of the available touring components or, for a smaller venue with rigging limitations, a cut down setup. Upon moving to the next venue and possibly changing the rigging/size/combination of the available touring equipment, all the tuning settings (levels and EQ) from the previous setup are not reset back to the factory defaults for the system as designed.

Without fully realising the dangers, the engineer starts tuning the system from the previous (sometimes forgotten or hidden!) delay, level or EQ settings that do not relate to either the new venue or the different setup configuration. This means the engineer is now adding more and more settings over the previous ones to attempt to tune the

Mixer

Measurement Mic Pink noise ref signal into mixing console as system tuning souce Mono summed mix of the main stereo mix is the Ref. (Ch 2) Input for the FFT software Ch Ou

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CONSOLE CONNECTS This connection method can be used at both the system tuning and optimisation stage of the process as well as while the show is actually running! Using this connection setup, the reference source (Channel 2 of the audio interface) for any FFT measurement is now derived from the mixing console instead of directly from the audio interface. All FFT measurements will therefore be comparing whatever is coming out of the mixing console — including the audio interface’s internally generated Pink Noise — to what the measurement mic is hearing coming into the FFT measurement input (Channel 1 of the audio interface).

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RECOMMENDED SMAART 7 SETTINGS

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These settings are a guideline for getting started with this tuning method. They may be refined, depending on your experience with FFT measurement software.

Smaart V7 in Transfer Mode with the Phase plot showing and Magnitude Scale at ±18dB. Averaging: 1

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Typical raw line array system response plot.

Phase Smooth: 1/6 Oct Mag Smooth: 1/6 Oct

system, and basing decisions on a completely false premise of what the system is actually doing.

Over the course of many shows in many venues this situation ‘creeps’ up on the engineer, and the system — now with masses of overused EQ and level settings — doesn’t sound anything like it was originally designed to.

With so many differing combinations of programmable system drive setups available (especially with line array-style systems) — not to mention how venues around the world can sound vastly different — IT IS VITAL THAT BEFORE STARTING TO TUNE ANY SOUND SYSTEM, THE ENGINEER BE ABSOLUTELY CERTAIN THE SYSTEM IS TRULY AT ITS FACTORY DEFAULT SETTINGS.

TUNING THE SYSTEM

STEP 1: ALIGNING MEASUREMENT SYSTEM

For the most accurate FFT measurement we always want the measurement microphone to hear the absolute minimum number of sound sources at one time. The ideal is a single point source. This avoids time/phase and excessive room excitation problems which seriously influence the accuracy of any FFT measurements. To begin make sure all other support systems except the Mains left and right are muted (including all the Subs).

Note: The Subs will be added into the total system during the final optimisation process but should be muted during all the original tuning processes. This will deliver the best possible tonal balance from the system itself without the Subs influencing all the low frequency measurement results.

Place the measurement microphone at ear height behind or next to the mixing console where the sound engineer will be creating the show mix. On the console, fully pan the audio interface’s input channel to whichever side of the stereo Mains the measurement microphone is closer to. Bring up the reference Pink Noise in that single side of the Mains L-R system to a comfortable listening AT 22

level. Using the level controls on either the audio interface or mixing console output, balance the mic and console output levels on the Smaart Magnitude screen so they closely match without any overloading or clipping. The resulting display on the magnitude screen should average somewhere around the centre 0dB line.

Using Smaart’s built-in Delay Locator function set the offset time between the direct sound from the mixing console (the reference source) and the sound arriving later at the measurement microphone (the measurement source). For accurate FFT measurements the two sources being compared must arrive as close as possible to the same time alignment. The accuracy of this time offset can be checked by looking at the upper phase display on the Smaart screen (unfortunately, explaining phase displays is beyond the scope of this tutorial). Some fine tuning can be done to the offset time but absolute accuracy will not affect the final result too much; so long as the ‘offset distance versus time’ reading in Smaart is somewhat relative to the actual distance of the measurement microphone from the source, including any additional possible system latency throughput time.

STEP 2: BEGIN TUNING ON THE MAINS

We are now ready to begin the actual tuning process. At this stage many vital decisions have to be made that will directly affect the sound of the actual show. Below 100Hz, the tuning is very subject to the style of music. For close-miked heavy rock shows, a +4 to +6dB step up in level from <125Hz (no higher!) and down is ideal, but for playback/speech-only systems, or orchestral systems with multiple open microphones at very high gain settings, a very flat response below 100Hz is required.

With pink noise exciting one side of the system and the levels balanced within Smaart, if you look at Smaart’s Magnitude display you can see an average of the raw response of the Mains system’s active side. Many factors including the rigging, pinning, original design of the system, design of the original factory crossover programs, and most significantly, the system’s interaction with the venue’s acoustics, will all influence what this raw response actually looks like. Regardless, the following approach to tuning any system should always be as consistent as possible.

Note: Even though we’ll be making tuning adjustments with only one side of the Mains system active, all the following adjustments need to be done on a group that has both the left and right sides of the main system assigned to it. All the following settings will then be actually adjusting both sides of the stereo Mains system.

Remember that what you are trying to achieve with this system tuning method is a system that accurately translates what is being created on the mixing console. When looking at the raw response of the system with known flat response Pink Noise as the reference source, it is immediately obvious what the active side of the main system is producing and how much it deviates from our stated goal.

A great many modern line array systems will ‘present’ at the raw response factory settings with a quite significant rising ‘tilt’ (boost) from the low frequency into low mid frequency response, and a long drop down in the high frequency response.

This type of response curve is, in many cases, pre-designed into those systems. If not corrected in the system tuning it will not produce the high definition and clarity required when using that system in a large acoustically-challenging venue. Using this FFT measurement setup we can immediately see that, if not corrected, that system is definitely not going to translate what is coming out of the mixing console accurately!

If this pre-existing low/low-mid frequency ‘tilt’ is not corrected, it will translate as a low/low mid frequency shelf boost on every input channel of the mixing console — even with the equaliser on all those input channels set to flat — ruining the nice ‘flat’ response of all those carefully chosen microphones on the stage.

STEP 3: BALANCING THE X-OVER Note: on some types/models of systems the crossover level settings may not be adjustable by the user. If so, skip this step.

EQ filters, when overused, can add unnatural artifacts to the sound. To avoid this, the first step should always be to balance the level relationship

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Levels Adjusted

Levels Plus EQ Adjusted

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Response plot after band level adjustments only.

between the low, mid and high bands of the system using the crossover levels. To avoid losing overall level, one of the three bands should stay set at 0dB and the other two bands balanced against that band. Set these band levels to get the raw response as close as possible to a flat line between 100Hz and 8-10kHz. Obviously there may still be some bumps in the response, as this is still the raw response of the system, but balancing the crossover band levels first will dramatically reduce the number of EQ filters required to get the system tuned correctly.

When in doubt about band level settings, where there are dips in the response always set the particular band level higher with the broad dips being closer to the reference line. It’s better to use EQ filters later to pull down the peaks rather than add EQ boosts to fix dips in the response.

STEP 4: ADDING EQ FILTERS

Once the band levels have been adjusted for the best balanced low, mid and high response from 100Hz up it may be necessary to add some EQ Filters to correct for the remaining anomalies in the system response. ONLY EVER CORRECT FOR THE LARGEST, MOST OBVIOUS RESPONSE ANOMALIES!

What we are looking for with our Smaart setup is the average response of the system falling within approximately ±4dB of the ideal ‘flat’ translated response through the system. A system that is overtuned to be absolutely ‘ruler flat’ just doesn’t sound musical and starts to sound far too clinical and unnatural. Due to the many measurement variables in a large venue, attempting to get a system tuned to ‘ruler flat’ is an unrealistic goal. Moving the measurement microphone five or six metres to the left or right, forward or back will quickly demonstrate that response variables in some venues can be quite dramatic. Looking for a realistic ‘average’ response is the only practical goal!

Use broad parametric EQ filters very sparingly to correct all the most obvious response anomalies through the full range of the system. WHEN USING PARAMETRIC FILTERS IT IS ALWAYS BETTER (AND MORE MUSICAL!) TO USE ONE BROAD Q FILTER CENTRED AT THE PEAK OF THE AREA TO BE CORRECTED RATHER THAN MANY NARROW ONES. Small very narrow peaks and dips

in the raw system response need to be treated very carefully as they may be false readings due to floor bounce or cancellations, etc. [See Ewan McDonald’s How Do You Measure Up? Flat-lining a PA tutorial in Issue 107 for more]. If in doubt about a narrow

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Response plot after adjusting only the major response anomalies with careful EQ.

response anomaly, leave it alone but try moving the reference microphone around to confirm if it truly is a real response problem. For the high frequency area of the system it is not a good idea to attempt to boost the high end of a system above 8kHz — let it roll off at its own natural rate.

For a rock music show, the frequency range below 125Hz (no higher!) can be allowed to step up from the 0dB point, forming a low frequency shelf in the response of approximately +4 to +6dB. Any wide peaks in this low frequency shelf region need to be adjusted very carefully as, due to reflections and other variables, the FFT measurement reading may be very inaccurate at low frequencies. It is often better to voice the system or use some known music reference material to adjust this area.

THIS LOW FREQUENCY STEP UP IN LEVEL MUST NEVER BE

The goal of this method is to have the sound system reproduce as closely as possible the same tonal balance and perspective — i.e. ‘the mix’ — generated by the operator at the mixing console’s output.

ALLOWED TO GO HIGHER THAN 100-125HZ AS IT WILL CREATE A LACK OF LOW/MID CLARITY DUE TO THE LOW FREQUENCY OVERTONES FALLING INTO THE CRITICAL 100HZ TO 500HZ MUSICAL INSTRUMENT FREQUENCY SUMMATION ZONE.

Using this method is the key to maintaining tight, controlled low frequency definition within the mix.

STEP 5: FINAL ‘VOICING’ OF THE SYSTEM

Now turn off the Pink Noise and play a wellrecorded ‘known’ music track into the Mains (only) of the system at a reasonable level. Looking carefully at the Magnitude display on the Smaart screen you should see a reasonably flat response above 100Hz indicating that what is coming out of the console is being reproduced on the system. It may be necessary to slightly modify some of the EQ filters you have added due to the fact you are now seeing the response of the full L+R Mains system.

Again, the goal is not ‘ruler flat’ but a good average flat response.

FFT software, including Smaart, is a great tool for tuning sound systems but as it relies on a measurement microphone placed at one position within the listening area for each measurement it does not ‘hear’ the full acoustic picture of the system in the venue nearly as well as the human ear does. As a double check, and to discover any extra anomalies between the system interacting with the venue acoustics, wire a favourite microphone into a completely ‘flat’ — ie. no EQ or hi-pass filter — input channel on the mixing console, and actually ‘voice’ the system you’ve just adjusted. A couple of good ol’ ‘Check one, twos’ will soon let you hear what a system that accurately translates the mixing

console’s output should sound like.

For some users new to this tuning method there will be a learning curve! If you’re not acclimated to the ruler flat sound of world-class recording studio control rooms, it may seem a little strange at first; light in the low end (the Subs should still be off!) and a little bright in the high end. Try not to change your previous settings between 100Hz to 10kHz too much but focus on the area below 100Hz and, with the board mic running into the full Mains system (L+R only, everything else still muted, including Subs!), adjust the EQ Filters to remove any overtones, room interactions, etc.

In many cases just a solid medium Q parametric EQ cut at 125Hz can make a huge difference to the entire overtone, interaction, definition and tightness of a sound system in a large venue.

Based on what you are hearing when voicing the system, go ahead and make adjustments that are as small as possible. Be sure to check that the left and right sides of the Mains system sound really close in tonality by panning back and forth between them. Once you are happy with the Mains, it’s time to start repeating these steps for all other support sections of the system.

STEP 6: TUNING THE SUPPORT SYSTEMS

It is important to realise that while the sound engineer at the mixing position will usually be listening to both the left and right Mains system — ie. in stereo — large sections of the audience AT 25

Ideal Rock ’n’ Roll Response

Ideal Orchestral Response

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Ideal final response plot for a rock music-style show after balancing and EQ adjustments.

may primarily hear only one smaller or single mono support system (eg. a Side Hang or Front Fill system). This forces us to tune the Mains (mix reference) system first and then, repeating the same tuning methods, individually tune each of the separate support systems, so that they all have reasonably closely matched ‘tonal balance’ characteristics. In the final system optimisation process all of the support systems must then be carefully level balanced relative to the Mains system to achieve a completely even, matched full system for the majority of the audience listening area.

Another factor common to many large system setups is that the various support systems use different types and models of speaker cabinets to the Mains left/right system. In their raw state they will not have the same tonality as the Mains system, which means these support systems must be tuned and optimised separately to match up as closely as possible.

Tuning all the extra support systems of a larger system uses exactly the same measurement setup and methods that were used to tune the Mains system. The measurement microphone needs to be moved to an area that is hearing a good overall average of what the audience will be hearing within each of the support sound systems’ coverage area.

Note: Whenever the microphone is moved, be sure to reset the delay offset time within the FFT software to compensate for the new distance from the sound source.

Again be sure to mute all other sections of the sound system except the one section the measurement microphone is meant to be hearing.

Starting with Step 3 (if accessible!) and repeat Steps 3 to 5 until all the additional support systems of the main system are individually tuned to sound as tonally matched as possible to the Mains system.

Note: Due to the danger of excessive low frequency coupling anomalies between the Side Hang system and the Mains front system (which will alter the perceived low frequency balance at the FOH position) be very careful when tuning the Side Hang system.

In many highly reverberant venues merely using a Side Hang system means we are now exciting a AT 26

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Ideal final response plot for an orchestral-style show after balancing and EQ adjustments.

whole different area of the venue which can create serious reflections and low frequency summation, along with various other problems that require much additional tuning/balancing adjustments.

To minimise these interaction problems when tuning the Side Hang system, make sure it does not have an excessive raised shelf in low frequency response below 125Hz. In fact, it’s a good policy to tune the Side Hang array to have a very flat low frequency response as, especially within the audience area where the Mains and the Side Hang system overlap, there will usually be more than enough low frequency energy due to the inherent coupling between those two systems.

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OPTIMISING THE SYSTEM

STEP 1: BALANCING & TIMING SUPPORT SECTIONS TO THE MAINS

Once the Mains and all the various support systems have been individually tuned the final optimisation process can be performed. The goal is that when the engineer is running the show he or she can be absolutely confident the vast majority of the audience is hearing sound closely matched in tonality and balance to what he or she is hearing at the mix position.

This process is a little complicated but, if done correctly, will not only balance the levels between all the various support systems as they relate to the L/R Mains system, but also create perfect sound imaging between those elements for the audience members listening to overlapping sections.

Unmute but turn down all the master levels for each of the support systems leaving only the Mains system on and set to the normal operating level.

With a favourite music track playing at a moderate level, walk to an ideal position between the Mains and the first support system you want to optimise, say one of the Side Hangs. Using a wireless tablet or with an assistant on a radio at the FOH control area, slowly turn up the master level of the section of the support system until it is perfectly balanced against the Mains system. Once you feel the balance is very close, walk back and forth through the whole area while muting and unmuting the support system to get a feeling for what it’s doing,

relative to the Mains system. Correct the level of the support system as necessary. THE AIM IS TO GET THE BALANCE AS SEAMLESS AS POSSIBLE.

After the levels are balanced, the next step is to get the ‘imaging’ between the Mains and support system correct. There are a couple of methods that will work well. The more exact method is to place the measurement microphone within the overlapping area between the two systems and perform a time delay measurement for each system separately then add an offset delay difference to the support system. Unfortunately this method is totally unusable outdoors with any wind present and takes time; which at most show setups, is not always available.

A quick method that can be reliably used under all conditions is to create a ‘click track’ on a CD or other playable media. If you don’t have a capable synthesised click track, tap a coin on a metal surface multiple times and EQ out any low frequencies. Loop it to create a long track of about 30 minutes. When played through the system it makes it remarkably easy to align almost any pair of sound sources by timing one source to another until only one distinct click is heard without any doubling/time offset effects present. This method assumes, of course, that when you are setting this ‘imaging’ timing you are listening within the overlapping area of the two systems. Right where you want the imaging to be at its best for the audience.

Repeat this process for all the other support sections of the total system until each and every section of the sound system is tuned, timed and balanced relative to the left and right Mains system.

STEP 2: ADDING IN THE SUBS Note: At this stage the Sub Bass systems need to be phase aligned to the Mains system. As the instructions on how to do this are beyond the scope of this tutorial, please refer to something like Bob McCarthy’s articles on aligning subs at www.bobmccarthy.com

Up until this stage we should have had all the Sub Bass systems muted to ensure the Mains and support systems sound as even and balanced as possible for the majority of the audience areas.

Sub Bass systems may be more efficient in certain audience areas and ruin our perspective of the low

frequency balance of the total system.

When setting up the level balance between the low frequency elements of the sound system and the Sub Bass systems, it’s vital that the Sub Bass systems only ever be set as an extension of the main system low frequency sections and not as a separate, louder entity. If dramatic amounts of low frequency ‘sub’ energy is required for certain types of music it should be added within the mix on the console, so it translates to any live recordings, broadcast feeds, etc.

To start, unmute but turn down the master level for the Sub Bass system and while playing a favourite track on the full system, slowly bring up the Subs master level until they become a true low end extension (only) of the system’s existing low frequencies. Mute and unmute the Subs to check what they are actually adding to the low frequencies and to hear if the room is handling all that extra low frequency extension. If the room is not handling it, then keep the Subs down in level as they may be making the overall system’s low frequency clarity worse rather than better in a difficult venue!

STEP 3: FINAL TUNING & VOICING OF THE SYSTEM

Once all of the previous steps have been finished and with the full system now unmuted, put Pink Noise through it one last time, then voice the full system a final time to carefully ‘touch up’ any minor tuning anomalies required to correct for the effects of the total system summation and interaction. Keep any of these final adjustments to a minimum otherwise all of the previous tuning work will be lost. The final optimisation process is the key to great, consistent-sounding shows in any venue with any type of system.

With a well-tuned system in an acoustically stable venue, while the soundcheck or show is running you should see a pretty flat line on the FFT system. If the line is constantly high in the high end, then the system tune is too bright; too low in the high end and the tune is too dull. It’s the same for the mids and the lows, although the lows below 100Hz may be very inaccurate (always too high) due to room decay, etc. Use your ears to decide if the low end is translating the low end of the mix from the console.

With this FFT setup remember you are only looking for an average of a flat line response. Don’t keep changing things based on one song which may have many peaks and dips based on that song’s particular texture — wait and check two or three different sounding songs and if something is constantly sitting out then adjust it slightly. Leaving this FFT measurement setup fully connected and with the measurement microphone next to the mixing console during the soundcheck and the show you can continuously see on the FFT display if the sound system is (on average) correctly reproducing what is coming out of the mixing console — the ‘mix’.

A Last Word of Warning! – To reap the full benefit of this tuning method and really hear the difference it makes, do not just carry over your old mixes/snapshots/ files from previous shows as they will inevitably have excessive amounts of channel EQ from earlier inaccurate system tunings (plus the ‘creep’ factor!). After using this new tuning method, you can certainly load up your old show file but it is vital you then put all of your console channel EQs back to flat and build a new show from there. As you build that new show I think you will be amazed at how much channel EQ you don’t need as you are now hearing the real sound of all your microphones!

AT 27

FEATURE

GEARING UP TO TAKE ON MONSANTO To record Neil Young at the famous Teatro, Jon Hanlon linked up the UA Green and Brown Boards with a Neve BCM10 and PSM12 into the ultimate DIY ‘large format’ console.

Feature: Paul Tingen

Artist: Neil Young & Promise of the Real Album: The Monsanto Years

AT 28

Neil Young an old fogey? With the legendary musician approaching 70, the description has been whispered a few times, even if there’s too much respect for the man for most critics to say it out loud. There’s a peculiar video on YouTube (Neil Young Shows Haskell Wexler His LincVolt) that initially seems to confirm the old-fogey angle.

Young — baseball cap, T-shirt and scruffy jacket, sunglasses, heavy sideburns and long hair — shows off his shiny, 1959 Lincoln Continental convertible.

His look has ‘old hippy’ pasted across it in neon, and the car itself screams nostalgia, suggesting an owner firmly rooted in the past.

The devil is under the bonnet though, because while it’s easy to miss when Young calls the car an “electric cruiser” right at the beginning, a moment later he explains that it’s powered in part by a generator that runs on “cellulosic ethanol, a future fuel made from waste.” The more cynical may still categorise it as an old-hippy pursuit — ‘Young’s an environmentalist, you know’. But as the video progresses and Young shows the gleaming, hyperadvanced technology just underneath the surface of the Lincvolt, the realisation dawns that the car is, in fact, totally and utterly futuristic.

The Lincvolt project is initiated and presumably funded by Young, and its mission statement is to “to inspire a generation by creating a clean automobile propulsion technology that serves the needs of the 21st Century and delivers performance that is a reflection of the driver’s spirit.” With the world heating up increasingly fast and mankind desperately needing to cut its CO2 emissions, it does not get more forward-looking and relevant to our times than that. Neil Young undoubtedly is an old hippy, but he also is far more with the times than many people a quarter his age. 1-0 to Young in his tussle with the 21st Century.

YOUNG STAR BUCKS

The same reflections, and conclusion, come to mind when considering Neil Young’s latest musical project, his 36th studio album The Monsanto Years.

Young’s 51-minute rant against the Monsanto multinational company (think Roundup), Starbucks, and big companies in general hijacking our democracies and endangering our environment and our lives, has come in for quite bit of criticism, ostensibly because the lyrics are too “didactic.”

These reviews also often have a hint of ‘who does he think he is to lecture us about anything?’ And yet, at a time when news of fast-approaching Armageddon is dominating newsfeeds everywhere, the question is far more pertinent why the vast majority of today’s artists take the ostrich-approach to the Big Issues Of Our Time. That’s Young 2, 21st Century 0.

There’s more. The Monsanto Years sees Young team up with a band of youngsters (’scuse the pun), called Promise of the Real, featuring Willie Nelson’s sons Lukas and Micah. Presumably, the idea is for Young to tap into their youthful energy and help him connect with a younger generation.

The album was recorded at the Teatro theatre in Oxnard, a coastal town half an hour north of LA, where Daniel Lanois set up shop in the late ’90s and

recorded and produced classic albums by Willie Nelson, Bob Dylan, and others.

The Monsanto Years was recorded in typical Neil Young fashion; quickly, nearly live, full of rough edges, and to analogue tape using vintage analogue gear. Yet it turns out that a Pro Tools rig running at 192k was also involved. So just like the LincVolt, a combination of vintage and advanced 21st Century technology was used, and the result sounds downright spectacular; big, panoramic, energetic, gutsy and very alive. All this surely helped in prompting sympathetic reviewers to state that the album sees Young, “at his usually defiant, belligerent and downright hostile best,” and “on angry, brilliant form.” 3-0 to Neil Young?

SHIP STIRRER

John Hanlon manned the ship during The Monsanto Years sessions, not only engineering, mixing and co-producing (with Young) the album, but also as the project’s general organiser, studio designer, and trouble-shooter. Perhaps it’s Hanlon’s old Navy and/or electronics background, but he’s extremely precise in his recollections, remembering that he got the first call from Young for The Monsanto Years project on December 17, 2014, saying that he wanted to record a new album… with Promise of the Real as backing band… at Teatro. Hanlon was immediately aware that Young’s simple pronouncement posed some significant challenges.

“It’s really important for Neil to find a space where he can set up and be comfortable,” explains Hanlon. “That usually means big spaces. Teatro is a big, empty theatre, with a high ceiling, and all the seats have been taken out, so the acoustics are cavernous. We could have worked in tons of places with better acoustics but Neil had his heart set on Teatro so it was my job to make the live area work and build a studio there for him. He wanted to record there because of the vibe and because of what the place represents. Particularly the great records that have been done there, even though it was with a different producer and 18 years ago.

“Also, the band consists of really accomplished musicians, with whom he’d worked at a benefit earlier in the year, and he wanted to work with them because they’re fearless and not afraid to go for things, yet take his lead. This meant that I had to record six musicians including Neil, which posed its own problems as we were working all-analogue with a limited amount of inputs and buses. I had a ton of work to do in terms of organising the acoustics, the band set-up and the studio. What I thought of Teatro did not matter. What was important was to make it work technically, and create an atmosphere where Neil can relax, and just be in the moment inventing and performing music. If I could achieve that, and Neil’s happy, I had a chance of recording great and heartfelt music. Because that’s what it’s all about, capturing the moment.”

THEATRE TREATMENT

Hanlon elaborated on the considerable amount of preparation that “capturing the moment” at Teatro required. “Given that Neil’s call came just a week

before Christmas, all I could do for the rest of the month was get on the phone and start lining up vendors, the acoustic team and so on. I started readying the recording space the first business day of the new year, January 5th. I laid down mats and carpets and put up gobos in the area where the musicians played to dampen reflections, and installed large panels against the back wall to break up the flutter echo. Of course, when the area filled up with gear it helped as well. The acoustic crew I had hired came in to work on the control room, which I had decided to build in the former projector room upstairs. It’s as bad an acoustic environment as you can have, with a big concave ceiling that was like the upside down hull of a boat, so we put clouds and traps up there, in the back and front, in the corners, and also left and right of the theatrical space, to turn everything into listening areas I could trust.

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

That’s the truest kind of nostalgia I can find

“The acoustic treatments were done in the first week, while I was installing the gear with Jeff Pinn.

Most of the gear came from Neil’s studio at his Broken Arrow ranch. I first worked there on his album Ragged Glory with producer David Briggs in 1990, using the Record Plant mobile truck. It was the first time I encountered the 12-input Universal Audio Green Board. It’s an all-valve console, built in 1965, based around UA 610 mic pre units with EQ at 100Hz and 10kHz. It sounds great, and was at one point owned by Brian Wilson. In all I had four consoles set up in the impromptu control room at Teatro. From left to right from where I was sitting they were a suitcase-model Neve PSM12, the Green Board, another 16-input UA board which we call the Brass Board, and a Neve BCM10 sidecar. THE BRASS BOARD IS A SOLID STATE VERSION OF THE GREEN BOARD, WITHOUT MIC PRES OR EQ, ALL HAND-WIRED WITH POINT-TO-POINT SOLDERING ONTO A BIG PIECE OF BRASS, AND ONLY LCR PANNING. I BELIEVE IT WAS HANDMADE FOR NEIL AROUND 1969.

“I also brought in a Pro Tools rig, tons of outboard, and Neil’s Studer A827 24-track tape machine, with a 16-track head block. I had used the same machine for the recordings of Americana and Psychedelic Pill (both 2012) but with an 8-track head AT 29

I had four consoles set up in the impromptu control room at Teatro… a suitcase-model Neve PSM12, the Green Board, another 16-input UA board which we call the Brass Board, and a Neve BCM10 sidecar.

block. Those albums were done with a four-piece, but I needed more tracks to be able to record six musicians. I set up 28-30 microphones at Teatro, which is not a lot to record a band, but you don’t use stacks of microphones when you’re confronted with the small amount of inputs that I had! I also set up a PA, mainly so they could hear themselves singing, and to amplify the percussion. My studio monitors were PMC IB2s as main monitors and PMC twotwo.6s as nearfields. I don’t EQ bottom end on small bookshelf speakers, and Neil wants playback to be as loud as possible when he comes into the control room with the band. So the IB2s served a dual function. The ability to check the low end is crucial for me, because the mid-range and the top are very affected by the low frequencies.”

CAPTURING IMPERFECTION

Setting up the Teatro recording space and studio took Hanlon and his crew two weeks. Once all the equipment and acoustic treatments were in place, he began the second phase of conducting the sessions.

“Neil had recorded demos of him singing with an acoustic guitar at Capitol Studios in LA, with Niko Bolas and Al Schmitt engineering. I brought a CD of that in on Monday January 19, for Lucas and the band to be able to hear the changes and melodies and lyrics when they came in for the first time. They also brought a few of their own tunes — it was part of the deal that I’d record them playing some of their own stuff as well. They ran through each of the eight songs on Neil’s demos, and a few of their own, over the course of a week, making sure they didn’t learn Neil’s songs into the ground, so to speak. Neil hates it when everybody learns things to the point that the life goes out of it. A lot of music today has AT 30

been perfected way too much, which is not human nature. Neil is into the human condition and into capturing imperfections.”

Having Promise of the Real run through the songs for a week also allowed Hanlon to perfect his setup and get the best sounds possible. THE PRODUCER HAD ONE MORE VARIABLE TO NEGOTIATE, WHICH IS THAT YOUNG PREFERS TO RECORD AROUND THE TIME OF THE FULL MOON. With the next new moon

on February 3rd he wasn’t expecting Young to arrive until the end of January, but in fact the main man turned up on the 26th. “Neil came in with one additional new song, and the band learnt that very quickly. We went straight for takes after that.

We usually recorded three takes of each song at the most. Sometimes we got it on the first take. If we didn’t have it in three takes, we took a break and moved on, then came back to the first song a few days later. The main thing is for everyone and everything to stay fresh.”

The recording and mixing setup that Hanlon had built at Teatro sounds straightforward enough, but the lack of inputs and buses meant a rather complicated web of signal chains. Using a 16-channel mixing desk without EQ or continuous panning, plus quirky mix preferences on the part of Young, required meticulous forethought. Hanlon went into detail on what was involved, starting with the microphones right at the beginning of the signal chains.

Hanlon: “The fact that I didn’t have many mic inputs was handy from one perspective, because the fewer microphones you use, the less phase errors you are going to introduce into your recording. I COME FROM A LOVE OF ENGLISH ROCK ’N’ ROLL RECORDS, AND ALL MY MICROPHONE TECHNIQUES

JOHN HANLON BIO John Hanlon first worked with Young on Ragged Glory (1990), as an engineer and mixer, and has since worked on a multitude of other Neil Young projects, including Weld (1991), Arc Weld (1991), Unplugged (1993), Sleeps With Angels (1994), Young’s Dead Man soundtrack for the Jim Jarmusch film (1996), Are You Passionate? (2003), Americana and Psychedelic Pill (both 2012). Having been trained in electronics in the Navy and worked in the computer business, Hanlon was hired by a small film sound post-production facility in San Francisco in 1973. He fell in love with tape machines and studio technology in general, played guitar, and later moved to LA, where he was a roadie for several well-known acts, worked as a studio tech at Record Plant Studios and A&M Studios, and eventually landed himself a job at the Beach Boys’ studio in Santa Monica. He went independent in the early 1980s, and did a lot of work with producer David Briggs, known for his pioneering work with Neil Young. The rest, as Hanlon says, “is history,” with much of Hanlon’s current time being taken up working at Young’s Broken Arrow ranch, just south of San Francisco, on the musician’s archives. In addition to his work with Young, Hanlon has over the years also worked with the likes of The Beach Boys, Cat Stevens, Dennis Wilson, Stephen Stills, R.E.M., Jackson Browne, and many others.

ARE BASED ON THOSE BY CHRIS HUSTON, ANDY JOHNS, EDDIE KRAMER — THE GUYS WHO CUT LED ZEPPELIN 2 — WHO ALSO USED A LIMITED AMOUNT OF MICS. My whole

concept at Teatro was to try to balance people in the room as best as possible, even before I put up any microphones or switched on the PA. I placed the guitar and bass amps in a semi-circle, with the drums behind them so the drummer is not getting the full force of the amps and hearing himself. As long as you maintain dynamics in the playing area, you get much better performances.

“I recorded the drums with only three mics, using the Glyn Johns method, with a pair of Neumann U67s above, at a 90-degree angle from each other and in front of the kick a Neumann tube 47 with a large piece of foam to protect the capsule from air pressure, and a Neumann 47 FET as backup. I had leakage from the guitars, but leakage is your friend.

You’re hearing everything at the same time, and that’s your record. I augmented that drum setup with a Shure SM57 on the snare and a Neumann KM84 on the hi-hat, but I only used them occasionally. I bused the kick drum to Track 1, and the 67s, SM57 and KM84 to Tracks 2 and 3.

“I had another Neumann 47 FET on the bass cabinet, and while I also had a direct, I usually used the 47, which went to Track 4 on the tape. I LIKE TO USE TWO MICS ON THE GUITAR CABS, A SHURE SM56 AND 57 — ANDY JOHNS STYLE — WITH ONE MIC CLOSE AND STRAIGHT ON, AND THE OTHER ANGLED. THE STRAIGHT-ON MIC GETS YOU THE MIDS AND TOP END, AND THE ANGLED MIC THE LOW END. As a result you don’t need EQ.

I used this technique on Neil’s Fender Deluxe and Magnatone amps, and on Micah Nelson’s Princeton, but Lucas played both my 1964 Fender Vibroverb and another old Princeton. Because I

didn’t have enough inputs, I had just a single 57 on each, angled at around 30 degrees off-axis, so I covered both the top and the bottom end. Neil played an acoustic guitar on the track Wolf Moon, a pre-war Martin D28 formerly owned by Hank Williams, and I recorded it with an AKG C12A and direct from the pick-up. The guitars went to Tracks 5 and 6, and Neil’s on Track 7.

“I had a Neumann KMS140 on Neil’s vocal, which is cardioid, because he tends to move around a lot, and this was recorded to Track 8. To pick up less from the room, I used hyper-cardioid Neumann KMS150s on the three band members who sang; Lukas, Micah and [bassist] Corey [McCormick]. These went to Tracks 11, 12, and 13. We also overdubbed backing vocals on some songs using a Telefunken 251 and a 47 FET for the double. I recorded those overdubbed vocals directly to Pro Tools, and they came up on Channels 15-16 on the Brass Board for the mix.

“Track 9 had percussion, which I recorded with a pair of fixed cardioid Neumann TLM103s.

THE ROOM MICS WERE ON TRACK 10. I PUT UP FIVE OF THEM, CONSISTING OF A PAIR OF KLAUS HEYNE-MODIFIED NEUMANN U87S IN FRONT OF THE BAND, TWO COLES 4038 RIBBON MICS BEHIND THE DRUMS, AND A TUBE NEUMANN 47 LOW IN FRONT OF THE STAGE FOR MORE BOTTOM END.

I didn’t use the Royer ribbon mic I put up on the balcony, because it had too much delay, which made it useless. It would have gone to Track 15.

“And finally, on Track 14 I had the subkick. I sent the kick mic out to a big subwoofer in the back of the room and recorded that back in, so I could get more bottom end on the kick. It excites the room and also gives the drummer a better sense of his kick drum. In the end it was a 14-track recording. I also had a pair of AKG C12As on Neil’s ‘Gold Rush’ upright piano, which is named like that because it was used on After The Gold Rush, and a pair of matched AKG 451s on a pump organ. I never used any of them. But if there are instruments in the studio that Neil can play, I better be ready to record them, otherwise I risk being in a world of hurt!”

INS & OUTS

So far, so straightforward, though Hanlon’s approach in reducing 28-odd mics to 14 tape tracks was not quite as clear-cut. Hanlon elaborated both on his bussing and some of the outboard he put into action. “The outboard was a lot of tube and old solid-state stuff, and it was all used during tracking.

They included a pair of Pultec EQP-1As on the bass and kick, and a Neve 2254 compressor on the bass microphone and DI. On the guitars I had Neve 32264 compressors and Lang PEQ2s — I like what the Lang does to the low-mids and upper bottom end on guitars. It sounds great. The mic pres on all the vocalists were Neve 1073s, and Neil’s vocal went through my Quad Eight compressor. I used three UREI 1176 compressors on the other vocalists.

“All my inputs came down to 12 on the Green Board, 10 on the BCM10, and external 1066 mic pres for the bass and DI, and 1073s for the vocals.

The Green Board’s 610 mic pres were great for guitar amps and room mics, and I used one line input for Neil’s vocal from the Quad Eight.

I DIDN’T APPLY ANY EQ ON THE MICS ON THE GREEN BOARD, BECAUSE THE CONSOLE SOUNDS SO OPEN AND BALLSY, AT 32

YOU DON’T REALLY NEED TO DO ANYTHING. The drums

and percussion and other room mics came in on the BCM10. The Neve PSM12 was there because I needed extra buses. It was used to combine the room mics that came in on the Green Board and on the BCM10. The Green Board only has four buses, but none can be multiple assigned. I needed to combine the room mics that came in on the Green Board and the BCM10, so I used their echo and foldback outputs to go to the PSM12, where they were blended into a single output going to Track 10 on the tape recorder. I didn’t need the room mics in stereo as I already had enough width from all the other microphones, particularly the drum mics on Tracks 2 and 3.

“THE STUDER WAS RUNNING AT 30IPS, WITH 5000FT REELS, WHICH GAVE ME JUST OVER HALF AN HOUR OF RUNNING TIME. I HAD TO DO SOME ‘HOT’ REEL CHANGES, WHICH WAS TRICKY! After recording things on

the Studer, I transferred them to Pro Tools at 24-bit/192k. The reason was that if you start running tape a lot, you begin to lose high end. It may not be discernible to most people, but the sound does change. I love tape and love rolling it back and forth, but we treated these 2-inch tapes as masters. The way we worked gives you the option of doing all your mix preparation in Pro Tools, and then later using time code to connect the tape recorder again to mix from the actual tape. We didn’t do that in this situation. The 14-track Pro Tools recording of each session came up on the Brass Board and I mixed on that.”

If there are instruments in the studio that Neil can play, I better be ready to record them, otherwise I risk being in a world of hurt!

“It’s a similar situation with the mix. Everything you do with Neil is a fight against time, because he doesn’t like waiting. The moment we had finished the transfer to Pro Tools and they came in to listen back, I was mixing the session live on the Brass Board and that was our starting point. That first playback with me running the faders — or rotary pots in this case — doing a mix better be going somewhere, because Neil is really into the vibe of what’s going down in the moment. He’s not afraid to decide that’s the final mix. While he’s also not afraid to bin a mix, when necessary, often he gets wedded to the first thing he hears. David Briggs told me a long time ago: ‘IF YOU DO A MIX FOR NEIL YOU BETTER MAKE IT GREAT, OR MAKE IT UNUSABLE BY RUNNING A 1KHZ TONE THROUGH IT, BECAUSE ANYTHING IN-BETWEEN HE WILL USE!’

Given that Young and Hanlon are self-declared, diehard analogue fans, the use of Pro Tools is a little puzzling, though the answer, at least on Young’s part, appears to mirror his championing of the hi-res Pono music player. “The reason to go to digital is practical,” replied Hanlon. “But when we use digital, Neil wants to go to the highest resolution available. We have done listening tests, and 192k sounds great, though it does depend on having a good clock. We have used the Apogee Big Ben, which I think is very musical, and I checked out the Antelope, Rosendahl Nanosync, Aardvark Aardsync and the new Pro Tools clock, and they’re all really good. WE DID A SHOOT-OUT AND WENT WITH

“I consider my initial mixes roughs, but Neil doesn’t like me to mess too much with them.

He doesn’t like it when things get too perfected.

Sometimes I was able to do another mix because you always try to improve on what you have done.

Neil has good ears, so if he liked the new mix better, that’s what we went with. Either way, all the mixes were done on the Brass Board, which meant almost exclusively adjusting levels. If EQ or compression was necessary, I had to use a plug-in in Pro Tools, which I did in a couple of songs, using the UAD 33609.

“The stereo mix went to my monitors, back to Pro Tools, and to an Ampex ATR 102, with quarter-inch tape running at 15ips. I like quarterinch tape, and I decided to make a change. I MIXED

THE GRIMM AUDIO CC1, BECAUSE IT SOUNDED THE BEST

TO ANALOGUE AND DIGITAL AT THE SAME TIME, AND LATER

WITH 24-BIT/192K AND THE NEW AVID HD I/O CONVERTERS.

WE COMPARED THEM. SOME SONGS SOUNDED BETTER

“I love the sound of tape, and these days it’s more of an effect. I used to like the lower output tapes, because you can get more tape compression, but with the high output tapes now you really have to work to compress it, and sometimes the tape machine electronics start to distort before the tape saturates! With rock ’n’ roll there’s so much harmonic distortion on everything, I like what happens when you put the whole band on tape, but if I was recording jazz or classical, I might go strictly digital, at high resolution. But up to a point it does not matter what I like. Neil likes tape, and then wants it recorded to hi-res digital as soon as possible. Once again, just like with the place where we recorded, it doesn’t matter whether he’s right or wrong, if that’s what he wants and I can make it happen, I have a happy performer and will get better takes.

RIGHT OFF THE QUARTER-INCH TAPE, FOR OTHER SONGS

ON THE CLOCK

THE DIGITAL MIX SOUNDED BETTER. I also transferred

the tape mix prints back to Pro Tools. We did some editing on a couple of songs at Fantasy and Shangri-La studios, then I took the digital mix files to Bob Ludwig who mastered the album. I like to be there for the mastering, particularly with an album like this that I co-produced, engineered and mixed. It’s all my fault!”

Hanlon laughed. Whether he made mistakes or not was probably not the issue most prominent on his agenda, in the context of working with an artist who sees capturing the moment and imperfections as an essential aspect of “recording great and heartfelt music.” Judging by the aural evidence, Young and Hanlon succeeded in their aims. That’s 3-0 to Young, with a little bit of help from his friends, amongst them, notably, John Hanlon.

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FEATURE

A bunch of Aussies, including engineer/translator Eric Coelho, travelled to Havanaâ&#x20AC;&#x2122;s EGREM to help unite Cuban and Jamaican music for the first time.

AT 34

Story: Mark Davie Photos: Lara Merrington

You can row the 100-mile stretch of open water between Jamaica and Cuba in two days; fly there in an hour. But these Caribbean neighbours are so culturally distinct, more than just an ocean separates them. You’ve got reggae, and you’ve got rumba. Dancehall and dub versus spirit-filled salsa. Each country’s culture is so vibrant you could almost hear the rhythms if you stood on either shore. But the opportunity to see how well they play together hasn’t really come up, until now.

After successfully building a bridge between Australian and Jamaican musicians with the Melbourne Meets Kingston album, Australian musician and producer, Jake Savona, decided to amp up the stakes with the Havana Meets Kingston project. This time drawing musicians from the two Caribbean music cultures into one studio and see what would come of it. American guitarist Ry Cooder went to Cuba in the ’90s and helped Cuban musician Juan de Marcos González bring Buena Vista Social Club onto the international stage.

Jamaican reggae has flourished ever since Aussie Graeme Goodall built the first commercial studio there and co-founded Island Records. But would the two be able to play to the same beat?

JAMAICA, MEET CUBA

Eric Coelho made the trip to Cuba as the project’s recording engineer. And the feeling of unknown was across the board. They had the budget; funded by a successful Kickstarter campaign and Australian Arts Council grants. They had the contacts; Savona had been travelling to and making connections in Jamaica for the last seven or eight years. His Cuban counterpart, Australian percussionist Javier Fredes, had spent similar amounts of time in Cuba immersing himself in its music and traditions. A film crew was prepped to document the whole trip, flights and studios booked, and a cohort of Jamaican and Cuban musicians willing to give it a go. But even as late as getting on the plane at Kingston airport, there was still a bit of hesitation.

Only one of the Jamaicans, guitarist Winston ‘Bopee’ Bowen, had ever made the trip from Kingston to Havana; as a child on a school choir trip — too long ago for the 60-year old to remember what it was like. There was uncertainty amongst the Jamaicans, recalled Coelho, “Like… are they going to let musicians into Communist Cuba?” But an hour later, customs officials were well-wishing them onto the streets of Havana.

CRACKING EGREM

The group had locked in a whole week at Cuba’s famous EGREM studios, the same place Buena Vista Social Club was recorded. Like most things in Cuba, it’s a relic of the ’40s and ’50s.“I don’t think it’s been cleaned or maintained since it was built,” said Coelho. “We didn’t even have enough mic leads. I used a couple of spare mic leads I brought

in my backpack. I had to try to fix some of the cables with old Canon XLRs using my Leatherman and a really old soldering iron with a terrible tip.”

Unable to contact anyone that could give him a state of play at EGREM before the trip, Coelho just had to cross his fingers and wing it. When he got there, the gear list looked like the pictures but it wasn’t all working. “The Amek Mozart console was out of operation,” said Coelho. “I had 16 pres on two old Focusrite Octopres, the silver face ones.

Trying to ride gains during recording was really tough because they were so scratchy. Before a take I’d quickly grab a pot and move it 10 times to try and loosen up any crackle and dust on it. I asked if they had Deoxit or isopropyl alcohol to clean them up. They were like, ‘Nah, sorry.’ We couldn’t even get paper. We had to write charts out on the back of our flight itineraries. I also used the eight pres on a Yamaha O2R console into some Digidesign 192 interfaces, then into Pro Tools.

“We had no outboard dynamics or EQ. The Octopres do have built-in one-knob dynamics, but they just weren’t suited. I didn’t want to compress a lot anyway, so I just kept it open and dynamic.”

But it wasn’t all doom and gloom. The state of EGREM’s upkeep was offset by three important facts, said Coelho, “We had the room, amazing musicians and a really great choice of microphones.

Everything else really didn’t matter.

“The main EGREM live space has wooden paneling everywhere that acts like diffusers. It’s got a beautiful ambience and decay to it that I haven’t really come across in any large live spaces in Australia. I’ve been to some of the larger orchestral rooms around Australia and they can have amazing room tones, but this had something different. It was bright without being too bright. It’s got that wood sound but it didn’t sound too dull either. It was just the right amount of decay and reverb you want in a room.

“The microphone choice is phenomenal. I had four Neumann U47 FETs to work with, six U89s, and a couple of U87s. Jake brought his personal U87, which we used as the room mic. He was pretty chuffed about that. Then the typical dynamic mics; a couple of EV RE20s, a couple of Sennheiser 441s and 421s, and Shure SM57s and a Beta 52.”

TALENT LINE UP

Leading the way from the Jamaican contingent were Sly & Robbie, legendary rhythm section who’ve backed everyone from Gilberto Gil, to Petere Tosh, Santana, The Rolling Stones, even Bob Dylan; lead singer of The Heptones, Leroy Sibbles, who was also a session bassist and arranger at Coxsone Dodd’s famous Studio One; and Bongo Herman, hand-drummer, percussionist and singer who’s performed with Bob Marley and Jimmy Cliff.

On the Cuban side was Barbito Torres, Cuban laúd (in the chordophone family, think guitars) virtuoso from the Afro Cuban All Stars and Buena Vista Social Club; Changuito, a hugely influential

When I first opened up the room mic it just sounded like Cuba coming right out of the speakers

Cuban percussionist; younger Cubans who make up the modern core of Buena Vista, and half of Havana, it would seem.

Coelho: “The musicians would recommend other musicians, and you just kept getting exposed to more amazing talented people that would want to get involved. Once they’d heard of the project, we were getting musicians from all around Havana just rocking up at the studio on a daily basis.

“They would recommend people that would specifically sound good on a particular song. Then we’d get them in front of a mic and ‘wow, that’s the one’. It was just these different tones and textures.

Beautiful husky vocals from this Cuban guy in his ’70s, younger rap vocalists, and a lot of the Spanish vocals from a Cuban artist there.

“We also reinterpreted some of the Buena Vista songs. They’re traditional Cuban songs that’ve been around for years; they’re an institution. Buena Vista and Ry Cooder popularised them and brought them out to the Western world but they’ve been a part of Cuban culture for years. We thought it would be good to pay homage to some of those traditional songs by reinterpreting them with a one-drop reggae, dancehall or rockers rhythm, with a dubby, heavy bassline from Sly & Robbie.”

TALK BACK TRANSLATOR

Of course, no one knew if these pipe dreams were going to make a good record, or if it was just going to be a big mess of conflicting Caribbean rhythms.

Coelho: “I still remember day one at the console looking over at Jake and he goes, ‘Alright Eric, fingers crossed, let’s see if this works.’ Then they start playing. Sly & Robbie lay down a nice tough, heavy reggae rhythm, and the Cubans start playing on top; Changuito on timbales, their Montuno piano style, and congas.

“It was one of those magical moments, within five minutes of them playing together it just clicked and we had goose bumps all over. We kept going and recorded 25 songs over that one-week period.

They just embraced it, there was no contention.

We were just facilitating something to happen and letting them run with it.”

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Savona had asked Coelho to come for a few reasons. For one, he’s credentialed. He’s engineered a lot of reggae, recorded The Strides and Kingfisher, Nicky Bomba and the Melbourne Ska Orchestra. Lately, he’s been working with Andrew Stockdale on Wolfmother’s upcoming album, in-between stints as a lecturer at SAE Byron. He’s also found a niche as a live sound engineer who’s not afraid to lather on the dubby effects. “These days I take an Apollo Twin with my laptop and a little Korg NanoKontrol surface,” said Coelho. “I just map feedback and filters to that instead of bringing my actual Space Echo.”

The other useful talent he possesses is a fluency with the Spanish language. Savona can’t speak a lick of it besides ‘¡hola!’, so was relying on Coelho to play interpreter between the Cubans and Jamaicans.

Coelho: “Even though there was a language barrier, the professionalism of the Cubans and the Jamaicans was remarkable; they made it work with music. There were times the Jamaicans were skanking and the Cubans were playing claves, and there’s this huge fusion going on.

“There are these 2/3 clave song patterns and the Jamaicans are saying to me, ‘The Cubans are pushing it’, over the talkback. I explained to them that BEAT THREE IS ALWAYS ANTICIPATED IN THE 2/3 CLAVE, One of the many none-too-shabby options from EGREM’s vintage Neumann mic locker that helped capture the legendary acoustic.

AT 36

SO IT’S ACTUALLY A LITTLE BIT LATE; IT’S HOW CUBAN MUSIC GOES. AND THEY’RE LIKE, ‘OKAY WE GET IT, ROLL THE TAPE AGAIN MAN.’”

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They made it work with music. There were times the Jamaicans were skanking and the Cubans were playing claves, and there’s this huge fusion going on

RUM KILLS HUMDRUM

While the sessions were starting to gel, the cultural differences weren’t lost on Coelho, the de facto translator. “It’s amazing how they’re so far apart politically and culturally, yet they’re two islands in the Caribbean only 90 miles apart. Cuba’s more traditional and based around a lot of religion, whereas Jamaican reggae is more socio-political. It was nice seeing how those two elements blended.

“The Jamaican musicians had a really strong work ethic, they were almost militant about it. That toughness comes through, making sure everything was tight and well thought out. Whereas the Cubans were just loose and sometimes there would be 10 of them in the control room behind me drinking Havana rum and smoking cigars. It was a party. Music to them was life, like breathing air; it wasn’t work at all for them.

“And then production-wise, if you listen to Jamaican recordings from the likes of Coxsone Dodd at Studio One, Lee ‘Scratch’ Perry and

King Tubby. They had a really tough and overemphasised kick drum and bassline, with all the ethereal dubby effects on top. Then Cuba is all about playing together in the same room with a nice vibe and room sound; they never liked to be in a booth.”

That was essentially Jake’s vision for the sound of the record, said Coelho, “‘I want the Cubans to sound Cuban, I want the Jamaicans to sound Jamaican.’ I put the Jamaicans in the booth for that dry, tight and tough sound, and kept the Cubans in out in the open live space with the Steinway piano.”

At one point he did try coaxing a few of the Cubans into a booth, but they wouldn’t have a bar of it, which was probably for the best, said Coelho, “The live room is a huge wooden hall, about 10 by 15 metres with really high ceilings. I captured the room so you open it up or pull it out to go with the arrangement. You can mute half of the Cubans and go really dry and tight for a tough Jamaican rhythm section. Then open it up AT 39

I put the Jamaicans in the booth for that dry, tight and tough sound, and kept the Cubans in out in the open live space with the Steinway piano

again for a chorus and bring in the percussion, the room sound and the piano. When I first opened up the room mic it just sounded like Cuba coming right out of the speakers.”

To demonstrate this EGREM phenomenon, Coelho played AT some raw recordings of two reinterpreted classics; Shingalin en Panama with Cuban rap vocalists, and Chan Chan, another Cuban traditional popularised by the Buena Vista Social Club. The room threw a warm blanket around the percussion, its natural predelay and short decay reacted perfectly to enhance the horn stab crescendos, and it turned the backing vocals into a cohesive ensemble. It’s the Cuban glue.

Soloing Sly & Robbie’s isolated rhythm section had an opposite effect; the dry, tight, tough and low sound of Jamaican reggae. Adding in just Bongo Herman and Bugsy playing a Nyabinghi pattern on the percussion, gave a completely different take on Chan Chan. There’s almost what feels like unlimited scope to jump between cultures in these arrangements.

VINTAGE CUBA

Coelho said he treated the session like any other, “The only difference was the music coming out of the speakers in front of me was absolutely stellar and amazing. I didn’t have anything to correct the performance with, and I didn’t need it. The musicianship was absolutely phenomenal, so I just had to stick a mic in front of it and hit record; don’t clip, don’t distort and you’ll be fine. I don’t want to AT 40

take any credit for how good it sounds.”

“I started by getting Sly’s drum kit sorted. We had the only hire kit we could get in all of Cuba.

They don’t have anything like a Billy Hyde’s hire company. Our man, Javier went all over Cuba to try and find this kit; it had a 17-inch kick drum that was 14-inches deep. Because there’s so many other percussion elements in Cuban music, the drum kit’s not as important as it is in Western culture and modern music.

“Getting an alright kick drum sound was quite challenging. We had a kick in, kick out, snare top but no snare bottom because we were trying to save on lines, a pair of overheads, two Sennheiser MD421s on the toms and an EV RE20 on the floor tom.

“We had Bongo Herman in another booth right next to Sly. He had one or two mics depending on the track; a close mic and an overhead to capture things like hand percussion and chimes.

“Bass was just DI’d, because we wanted to keep it separated from the Cubans. The guitar was also DI’d. We had the only Fender Twin we could find in Cuba but it was 220V… Cuba runs on 110V. Because all the Cuban percussionists were in the live room, the Fender Twin wasn’t going to be a good option anyway.

“The rest was congas, timbales, a whole array of Cuban hand percussion, and a second Cuban drum kit setup in the live room that had two overheads and a kick.

“We put a spaced pair of U89s over the piano.

The room mic was on a large counterweight mic stand about five metres off the ground and set to an omni-directional pattern. It was behind the piano, but capturing the rest of the room. The piano wasn’t as loud as the percussion elements, so we tried to balance it that way.

“We always recording the room mic with every overdub take. It’s the glue that gave us that Cuba EGREM sound. I’ve grouped the room mics in the session so I can mute them all at once; it flips between Cuba, no Cuba.”

NOTE IT DOWN

It was a strange situation, to be spoilt for choice when it came to vintage mics but not having enough lines on the interface to capture them all.

Coelho had enough mics to leave them set up over each instrument, but he’d “have to repatch depending on what we were doing next.” But like he said, it was all about those musicians, who by all accounts were phenomenal.

Coelho: “We finished each song as we went because often songs were thrashed out then and there in the studio. Sometimes all the Cubans would stand around the piano singing all the instrumental parts, and someone would be jotting them down with a notepad and pen. Then they’d all go to their instruments, I’d hit record and they’d nail it, first take. They were an amazing caliber of musician.

“Jake stayed on in Cuba afterwards and used his U87 with an Mbox and laptop to get a few other

While esoteric mics were a dime a dozen at EGREM, trying to find a simple fuse in Cuba proved problematic. Coelho had to rig up this little makeshift jobby with a bit of wire so the keyboardist could keep hammering away on his Nord.

little sound bites and vocal takes. He also used a couple of other small studios, but 90% of the recordings were done in those seven days in EGREM.” THE JOURNEY CONTINUES

The project won’t be finished till early next year. Over the next couple of months Jake will be recording more Jamaican vocalists in a blend of english and Jamaican patois. The plan is to include both older, more traditional Jamaican artists like Sibbles with some up-and-coming fresh talent. “Jake wants to break some younger guys, but we’ve got aspirations to get someone like Damian or Steven Marley on board to do some guest vocals on some of the songs.”

Similar to the line of Cubans at EGREM’s door every day, the one thing Coelho knows they can rely on is the insatiable Caribbean appetite for music. “While I was in Jamaica, we went to some of the reggae and dub parties, and there was always a line-up of vocalists trying to get the mic.

It’s the traditional dancehall culture that Jamaica is known for, just a DJ, a mic, and ‘toasters’ chatting or singing on the mic one after the other.”

FEATURE

Mitch Kenny got over a quick bout of self-doubt to mix Hermitudeâ&#x20AC;&#x2122;s Dark Night Sweet Light into a nomination for ARIA Engineer of the Year.

Story: Mark Davie

Artist: Hermitude Album: Dark Night Sweet Light

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LAYERING KICK DRUMS Mitch Kenny: “The three most important things on a record are the kick drum, the kick drum and the kick drum. I knew it before I went to The States, but making a lot of dance and hip-hop records really reinforced that idea. Even with that big L.A. rock sound, the kick drums are just so big and important. I’ll layer kicks till the cows come home.

98% of what I get to mix is done on laptops. I’m not getting stuff to mix that’s recorded in studios, which I’m completely fine with. When it’s being prepared in less than ideal environments, some people have got no idea what phase is, let alone where it is. They can make it knock in their room but don’t know the bottom end is out of phase.

“I’ve got a few kick drums that all do a certain thing. I use a live kick drum sample, and a 9th Wonder kick together quite often. Some of them might not be overly loud but it’s about making it work in different formats. Hermitude’s record, in particular, has to work in a club, on radio and on a phone. The very low bottom end can’t be too much for any of those three mediums, and I need at least one kick drum that’s got some front — something above 3000Hz — and is in the centre.

“I know I need to give-up and walk away quickly when I’m a long way through a mix and want to go back and mess with the kick drums again. Once I get them settled, that’s my benchmark and everything else fills in the space around that.

“I get a lot of records to mix with loads of stereo kick drums. I need something in the middle to hang the hat on, so I’ve got a really good collection of modernsounding mono kick drums.

“I’ll place kicks in by hand with the Tab to Transient command in Pro Tools. Before I start replacing, the first thing I do is check the phase on the kick drums I’m replacing, including the breaks and samples. After that, I check anything I’m adding.

Mitch Kenny should be in hospital. When I catch up with him in his temporary studio on Level 7 of a George St low-rise, he’s still occasionally coughing up the remains of a lingering bout of pneumonia. But there’s no stopping Kenny, which is how he got ill in the first place. He talks non-stop for the next three hours, rides his scooter across town to Sydney’s Studios 301 and gabs on again in front of a crowd for another 30 minutes.

He’s just finished up a five-month long stint, flying between ABC studios in Sydney and Melbourne to record and mix the music for a new TV show called The Divorce, starring Lisa McCune, Marina Pryor, Hugh Sheridan amongst a cast of other singing and dancing actor-types. Kenny describes it as being “like a Bing Crosby movie, where people spontaneously burst into song”. The ensemble he’d been recording was the Australian Opera & Ballet Orchestra. It’s a world away from Hermitude’s Dark Night Sweet Light — for which he scored an ARIA Engineer of the Year nomination, and the record we’re supposed to be talking about — but Kenny’s used to hopping codes.

“On the 808 in the verse of The Buzz, I automated the level to turn down the front of each kick. The front of it was just a little bit too clicky. Initially I tried to fix it with the SPL Transient Designer, but I couldn’t get it to work. It’s unusual. Typically I’m trying to find as much front on an 808 as I can.

“I then ended up using the Transient Designer to extend the body to go from hit to hit. I love that 808s can be found on everything, whether it’s dance, hip hop, or pop.

I even did a full-on, overblown Nashville country record for American Idol winner Scotty Mcreery — you can’t get more white-bread than that — and there are 808s in there just ticking along playing the tonic and dominant.”

CLASSICAL TRAINING HELPS

Even though he quit his music performance degree at Melbourne’s VCA, classical music has been good to Kenny. There aren’t many dedicated engineers who have tried finding their place amongst a deck of cellos. When he dropped out, Kenny did a stint at audio school and a bit of live sound work before ending up at Studio 52, where he really learnt the craft. Through his VCA connections, Kenny also freelanced his way through his fair share of classical music sessions.

Kenny had been flying back and forth between Melbourne and The States for a few years before he decided to head over to the U.S. permanently and try his luck. His mate, Jared Scott, had been engineering there for a few years, so on a wing and a prayer Kenny bought a one-way ticket.

When Kenny arrived, he soon realised he had to pass the critical eye of one of three women to get into the studio system: Paula Salvatore, Studio Manager at Capitol; Rose Mann-Cherney, the now-retired President of Record Plant Studios; or

Candace Stewart of EastWest. “Those three women controlled the whole movement of engineers,” said Kenny. “Especially staff engineers in the studios in L.A. They’re very good friends and they know each other’s needs.”

After walking around town introducing himself to every studio in LA, he finally landed a meeting with Paula. She knew a bit of his work, but with no positions available, directed him to go and see Rose. Without Kenny knowing — in the time it took to get from Capitol to Record Plant — Paula had already rung ahead to recommend him.

Apparently another interview wasn’t required, because by the time he arrived, Rose had already organised a walk-through with Jason Carson, the studio manager.

It was post-GFC, and Record Plant was the only studio that hadn’t lowered its rates. “The only artists booking the studio were rappers who liked being able to say they were working in the most expensive room in town,” said Kenny. “The positive side of that is they kept going and it was a really interesting place to be. The negative side AT 43

SIDECHAIN YOUR WAY OUT Mitch Kenny: “I ended up sidechaining the percussion break in Through the Roof with Waves’ C6 multiband compressor to lose all the bottom end in the drum loop and make space for the kick drums. When the kicks weren’t hitting I wanted to have the rest of it popping through, so a normal sidechain would have just taken away too much information.

“I used the same trick for the snare. It was hitting at the same time as the kick drum, so I sidechained the multiband’s top end to compress when the kick drum hits. It means they work together rather than getting in the way of each other, and gives me more space.”

CHANGING YOUR IMAGE Mitch Kenny: “I get a lot of mixes supplied where every track is a stereo file, which means you need to do something to manipulate the image. I tend to pan things in the centre or hard left and right, then I’ll stereo expand beyond the edges to get things out of the way or make a highlight of them. It’s why I’m so adamant about having a strongly centred kick; it helps balance that super wide stuff.

“I’ll very rarely tuck something in. It has to be doing a specific function. The reverb and stereo synths take care of that anyway by being all over the place in terms of the image.

“I find the trick is not about how wide you can make a track, it’s more about getting it moving. An example was automating the Aphex Exciter plug-in, the stereo width, and volume, simultaneously on the synth hook in The Buzz. There’s a moment at the end of the hook where it jumps up an interval and wants to pop out. It was supplied as a stereo sound and because I wanted it to leap into the middle, I automated Waves’ S1 Imager to the opposite extreme and automated the Aphex Exciter so it excites at the front of that note.

“All of a sudden, this part that was bouncing on the edges of the mix gets slammed into the middle, pushed up in volume, and distorted by the brightness of the Aphex Exciter. It’s exciting.

“When the mix is working for me psycho-acoustically, it’s like a diamond. I see the height as the frequency from low to high, the width is the stereo image and the front and back is basically the balance. When it starts to go up and down, forwards and backwards, and side to side all at once, that’s usually when I’ll send the mix off.

was that house engineers only ever dealt with one channel in and two channels out; because they were only recording vocals over beats. As I had a ‘proper’ engineering background (we all have to do everything in Australia), Jason asked me, ‘How are you with engineering a string session?’ My response was, ‘I eat string sessions for breakfast!

There’s fewer microphones than a drum kit.’”

The first session Kenny engineered in the US was a string session for a Mary J Blige record.

That classical music background paid off again.

Over the next few years, Kenny engineered a boat load of hit records including some for Elton John, Beyonce and Chris Brown. One of those, the No. 1 Don’t Hold Your Breath for Pussycat Dolls’ Nicole Scherzinger, put him in the room with hit AT 44

songwriter Billy Steinberg and Jimmy Iovine’s righthand man Dave Rene. Rene had a young Russian kid in the wings who needed an engineer… his stagename was Zedd. Kenny ended up engineering much of what became the mega-hit Clarity.

MAKING TRAKS

Kenny eventually decided to move back to Australia a couple of years ago, and again, his classical music background paid off. Although sessions didn’t come easily at first, he did end up recording the Australian Chamber Orchestra at Studios 301 for a Qantas session with Daniel Johns.

These days, Kenny is heavily entrenched with Elefant Traks artists Horrorshow and other hip hop artists from the One Day collective, which includes

Joyride and Spit Syndicate. He’d also previously worked with Angus (Gusto) Stuart from Hermitude when he and Urthboy produced the Paul Kelly cut of the Hunters & Collectors tune True Tears of Joy. “I’ve only been in Sydney for a short while,” said Kenny. “So it’s been nice how one thing has led to another, to another. I hope that means people are happy with what I’m doing.”

So tight knit is the hip hop community, Kenny actually got the call from Urthboy, akaTim Levinson, to ask if he wanted to do a spec mix for Hermitude’s new album. They’d had the track mixed by someone overseas and were hoping for a better result. “No problem,” he said. “I got the email chain of requests and had a relatively good idea of where they were going.”

PULLING IN YOUR TAIL Mitch Kenny: “I love TL Space’s emulation of the AMS RMX16 and Avid’s D-Verb, usually set to Medium Room 2. The two parts of my world — pop and classical — influence each other greatly.

My take on reverb tail lengths is heavily influenced by the classical side. It’s one of the things classical music producers are sticklers for: making sure they’re not only balanced in the image, but that their lengths are right for the music.

“The start of a sound is more important than the end of it. You get body and width from the end of a sound, but people aren’t going to tap their foot along to the end of a reverb tail. So I don’t want the tails to be so long that they’re getting in the way of the next sound. I want them to be falling in the pocket.

“That said, reverb tail lengths change all the time and can really date a record. Listen to Pour Some Sugar on Me and Mutt Lange has the reverb tail louder than the vocal! It’s amazing, and exactly how it should be because it’s a moment in time. Even from this record, which was mixed in February, to the end of the year, snare tails have shortened up a bit.”

VOCAL THROAT Mitch Kenny: “I didn’t record the vocals on this album but I did learn a lot about how to record them from working with Kuk Harell, who produces vocals for Beyonce, J-Lo, Bieber and the like. The biggest lie anyone tells a vocalist is, ‘one more.’ He will track through until he’s absolutely convinced he has enough to comp, then he will say, ‘four more.’ That way the artist knows you’ve got everything you need to make a vocal, and the ‘four more’ is because we might stumble upon a happy mistake or a little bit of magic. Because they know you’ve got it, they relax and might find a bit of brilliance. Then after four, move on. I can make a barking dog sound like Pavarotti with Melodyne, but you can’t manufacture emotion.

“I put the Antares Throat plug-in on Young Tapz’s vocal hook in Through the Roof. It does a couple of things. It scoops out around 200Hz and — this is from someone who’s recorded far too many vocals — makes it sound like there’s more top palette in the sound. It also makes it sound more present. I don’t change the ‘throat’ and ‘glottle’, but I change the breathiness.

This is adding 12 — whatever that means — at 4000 cycles, and pushes things out a little.

“I also automated McDSP’s Futzbox in at the end of the hook, because with everything else coming in, it just made it sit better. Overall, I couldn’t depart too much from the sound of the pre-effected chopped up vocals, because they were switching between them and the hook quite rapidly.

DON’T DELAY TOO LONG Mitch Kenny: “I’m a stickler for automating delays rather than leaving them in, because they take up so much space. I’m a big believer that the ear picks up changes in tone more than changes in level. I love using Waves’ H-Delay for its control, but hate the ‘analogue circuit’ part of it; it just adds noise. When you compress it, it just keeps bringing up the noise.

“I automate my delay throws with a physical fader to catch the nuances in a more musical way. For instance, on Young Tapz’s vocal for The Buzz, I didn’t want the ‘z’ sound at the end of his hook to repeat in the delay. So as well as band-passing the delay signal, I quickly faded the send in and out to catch just the middle of ‘buzz’ which helps it to sit better in the mix.”

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MASTERING THE MAGIC FADER Kenny’s setup is relatively simple. He’s a big exponent of mixing in-the-box. He holds nothing against consoles, having developed his craft on them, but he’s fully aware that the ability to recall is hugely important to his clients.

Mitch Kenny: “Tal Herzberg, who unfortunately died so early, was my walking reality check. When I got to The States I was all console. He shook his head in his stern way and said, ‘When the A&R man rings you at 11 o’clock for a change and you’ve done the change while he’s still on the phone justifying his decision; and you’re sending it to him as he’s still arguing and telling you what the change is; you’ll never mix on a console ever again.’ He’s right. The reality is the amount of changes we need to make and the deadlines that we work under most of the time are completely unforgiving.

“If an artist asks for a 0.1dB increase on their vocal channel. I can’t hear that, but if it makes them feel better when they’re listening to the track a year from now, I’m completely okay with that.”

Kenny has an Apogee PSX100 two-channel converter plugged into his Black Lion-modded 002 via S/PDIF. He uses the Apogee to interface with his analogue master bus chain, which comprises a GML 8200 stereo parametric EQ and a Smart C1 compressor.

MK: “There’s bandwidth for days on the PSX100 and it does enough that the difference between it and splitting everything out on an SSL K is so minimal that it’s not worth it in my opinion.

“I always have the C1 master bus compressor on. It does the lion’s share of the compression for the track, especially on the kicks. It also lets me know when my kick drums are loud enough.”

One key part of Kenny’s mixes is his use of the ‘Magic Fader’. On Through the Roof

he pushes the master fader up at various points to increase impact, and slowly brings it back to unity before the next hit. The C1 compressor keeps the overall level from fluctuating wildly but by pushing the level into the compressor it gives Kenny a controlled level of momentary distortion that adds to the excitement and movement of the mix.

MK: “Not that long ago, I tried to mix something without an 8200 and my bottom end was wrong. I always start off with a shelf at 30 cycles taking 3dB out. Then if I don’t have enough, I’ll turn it off. It’s a format thing. You don’t need too much information below 30Hz because it’s so over-accentuated in the clubs anyway. It also translates better for radio because you don’t have all this extra information that has to be compressed through a transmitter.”

LISTENING WITHOUT LISTENING I can make a barking dog sound like Pavarotti with Melodyne, but you can’t manufacture emotion

When he opened Gusto’s reference mix, he had a moment of self-doubt. “I spoke to Joyride about it, who I write with,” said Kenny, “because the first time I opened it up, my initial gut feeling was, ‘this sounds amazing. What do I do to make this better?’

Joyride’s response was, ‘Just do your thing.’”

So Kenny got stuck into bringing his particular mix approach, but in an über-aggressive way, and that one spec mix turned into the entire record.

Still, after he’d mixed five or six songs without any feedback, the self-doubt crept in again. “I spoke to my manager Bernadette about it,” said Kenny. “Her response was, ‘If they didn’t like it, you’d know about it.’ One day Dubs [Luke Dubber] showed up at Studios 301 with some more parts for me. I said, ‘Mate, I’ve got to ask, are you happy with how this is going?’ And he says, ‘Yeah man it sounds sick!’

So I just kept doing what I was doing.”

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Mitch Kenny: “While I’m flying the kicks around, it gives me time to listen to the track passively. With it just washing over me, sometimes I hear things I normally wouldn’t. I can hear the macro and the micro at the same time. I think I do better mixes when I’m distracted and not so actively listening. I don’t know if this is true or not, but someone told me they did a study on the brains of mix engineers and found they could slow their alpha waves down quite a lot. The only other people that could get their brains to that sort of state had been engaging in meditation for a long time. I believe it.”

Besides the lack of early feedback, Kenny thoroughly enjoyed mixing the record. “In terms of two humans to make a record with,” he enthused, “I’d make a record with them any day of the week.

They’re super-talented guys, lovely as people, really pro, and they work hard.

“Gusto has a really good understanding of actual engineering too. Things came with headroom, were labelled correctly, and he supplied the dry parts alongside the printed effects.”

Hermitude are an incredibly creative duo.

Through the Roof’s horn parts sound every bit like they’re sampled from inside a genuine Cuban nightclub, but they’re actually programmed and effected. The pair also went out and recorded a lot of sounds to build their tracks with: sizzling bacon was used as a substitute for white noise to give a little extra flavour; they struck pipes in tunnels to

augment drum hits; and got girlfriends to make whale noises. They’re no slouches in the playing department either. Dubs flexes his keyboard chops on the song Metropolis, which they dubbed the ‘NAMM solo’ because it sounds like a hired gun ripping on a trade show floor.

At the end of the day Kenny just wanted the record to sound like a Hermitude record, because “Hermitude gets judged by this. I don’t. Most people won’t even buy the record as a physical format. Besides, with how hard it is to find credits and how poorly they’re represented online, it’s a joke anyway. Joe Bloggs from Dandenong doesn’t care who mixed the record, there’s only a very small group of people who judge me. Hopefully those people like it but if they don’t that’s fine too, because it’s not about me, it’s always about the artist. Always, always, always.”

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FEATURE

Story: Paul Tingen

Artist: Macklemore & Ryan Lewis Song: Downtown

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ABOUT JON CASTELLI Originally from New York, Castelli studied saxophone at the Hartt School of Music on Long Island, and started and ran a studio together with his father in the same area, where he cut his engineering and production teeth.

In 2013 he signed to Mirrorball Entertainment in Los Angeles as mix engineer and producer, where co-founder and star mixer Tony Maserati took Castelli under his wing. Castelli: “The first thing we mixed together was a Lady Gaga song, then he invited me to move in across the hall from him, where I still have my studio. It’s all in-the-box, my only outboard being a Rascal Audio Two-V preamp and a Kush Audio UBK Fatso compressor, both

Bruno Mars and Mark Ronson’s Uptown Funk was a very well orchestrated pastiche of ’80s funk. All the key elements were there — keytars, horns, low fisheye lenses and squareshouldered pastel sports blazers. But there was still a collected, serious side to the parody. Whether it was yammering on the phone while getting a perm or reading the paper while their shoes were shined, Mars and Ronson were getting down to the business of being funky.

Macklemore & Ryan Lewis don’t have to parody jack. Their songs and videos are too oddball to attract direct comparisons. Especially so on their latest hit, Downtown, which, on the face of it, could be read as a direct response to Uptown Funk, but is so much more bizarre.

Downtown’s chorus — top-lined by Foxy Shazam singer Eric Nally while riding a chariot pulled by Royal Enfield motorcycles — doesn’t come in until almost the two-minute mark. The verses — oneupping the anti-cool vibe of Thrift Shop by geeking out on mopeds — are made up of completely different sections, featuring not only Macklemore, but also rap pioneers Melle Mel, Kool Moe Dee and Grandmaster Caz. The song has echoes of both the ’70s and the early ’80s funk aped by Uptown Funk, and is almost entirely played on live instruments with prominent sax, trumpet, tuned percussion, and piano.

of which I use over the master bus. I also have two of Tony’s Neve 1066 mic pres, plus the Black Lion Microclock MkIII and Sparrow AD/DA. My monitors are PMC IB2s, with a Bryston 4B amp. I also have Chris Pelonis Mode 42 MkIII speakers with concentric drivers that I use like Auratones. They are awesome.”

Castelli’s credit list already includes mixes for Lady Gaga, Beyoncé, Ariana Grande and Backstreet Boys. As one half of 4FRNT, he also did a remix of Somebody That I used To Know that was endorsed by Gotye. As well as Downtown, Castelli eventually mixed the entire yet-to-be-released Macklemore & Lewis album.

The song’s unconventional nature apparently also initially foxed its makers. Mixer Jonathan Castelli recalled, “Ryan called me and said they had been working on this song for two years, and they really needed fresh ears and perspective to help finish it.

When my assistant, Ryan Nasci, and I listened to it for the first time, it sounded crazy. It was going to take a lot of work to make it feel like one cohesive concept. It sounded like four songs stitched together with transitions that made the sections sound completely different. There were no effects and the voices weren’t cutting through enough.

The challenge was to make it sound like one whole piece so America would not be freaked out by it, as well as getting the voices to step out in front, make the whole song jump out of the speakers, and take it into the radio world.” A big job for an already unwieldy song.

GETTING DOWN TO IT

Castelli got the gig through Joshua ‘Budo’ Karp, a frequent collaborator with the Seattle duo he’d met at a writing session a few years ago. 18 months after that first meeting, Budo sent Castelli a text asking if he was up for mixing some Macklemore & Lewis material. He was hardly going to refuse, and got stuck into the mixes at his Mirrorball studio.

The uncertainty around exactly how to approach Downtown surprised Castelli. It was the fourth

song he’d mixed for the album and the direction for the first three had been fairly clear. Although the duo weren’t exactly clear about the direction for Downtown, they did recognise that Castelli’s first mix wasn’t what they were after.

Castelli: “For my first mix, they had given me no direction, and when I sent it to them, I was very happy with it. Ben [Macklemore] contacted me and said, ‘Man! How did you get Eric’s voice to cut through in such a loud arrangement?’ But when Ryan [Lewis] called he said ‘Jon! I really like that mix, but it is completely the wrong direction! I like that warm, fat, more tube-like sound with extended bottom end, that you got on the other songs, but this song needs to be a little more natural sounding.

It needs to be less EQ-ed in the low end, with more of a natural mid-range — more solid state.’

“I immediately knew what he meant. I love tube gear and tend to go for big fat bottom with subharmonic stuff and saturation. That worked on the first songs I mixed, but for this song he said, ‘do yourself a favour and go listen to Grandmaster Flash’. Finally he gave me some of the vision. If he’d given it to me earlier, I could have saved myself three days’ work! I had worked really hard on it, because I knew it was earmarked to be the first single from the album. Usually you can backtrack in a mix, but not with this song. I decided to start again. Downtown eventually turned into a two-week mix!”

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DOWNTOWN MIX The Pro Tools session for Downtown is an impressive 112 tracks, including group and effect tracks. From top to bottom it consists of five mix tracks in purple, including the rough mix right at the top, eight group tracks in bright green (drums, guitars, keys, synths, strings, horns, vocals and effects), 34 drum and percussion tracks (mostly red and yellow), three effect tracks, five bass tracks (mostly orange), one piano track, seven guitar tracks (dark green), one Oberheim synth track, and five horn tracks (yellowgreen). The bottom of the session is taken up with a massive 45 vocal tracks, including 14 for Ben Macklemore (though the three tracks labelled Mod are of Macklemore demoing guest rap vocals), 18 tracks for the three guest rappers, Eric Nally’s, the choir, and other incidental vocals in between.

While 112 tracks is indeed large, Castelli’s mix session actually comprises the stems Ryan Lewis’ engineer Stephen Hogan supplied. Hogan had bounced them out of an original session that added up to well over 200 tracks.

TWO-WEEK MIX

Two weeks for a mix sounds exorbitant, but Castelli said his mix for Downtown required a lot of finetuning because, “Ben and Ryan wanted to come out with a song that was unlike anything that had been released in the last 10 years. They’d had three No. 1s, and although they want to be successful, they really wanted to ‘wow’ people. High chart positions were less important. During one phonecall I said to Ryan that I saw the song as their response to Uptown Funk. Those guys are posh and living in uptown, whereas Downtown is more down-toearth. He really liked that perspective.

“Specifically, instead of going for a big bass, R&B sound as with the first three songs, my challenge was for Downtown to sound more like a throwback to the seventies; more Neve and API than tube.

Of course, it still needed enough bottom end and punch to sound current. The problem with my direction at that stage was more sonic, as I had already to some degree resolved the issues of AT 50

Downtown — with rap verses and a clean chorus that transfers into something akin to classic rock — sounding very segmented. And, as Ben had indicated, I had managed to get the vocals to cut through. Bringing all that together was a big challenge. But I love being challenged!”

With so many pieces to assemble, more than ever, Castelli just took the time to listen. He “just listened to it for a good hour, soaking it in. For me mixing is about the journey and emotional connection you get with a song, not about what plug-ins I use. With other songs I might dive in right away, perhaps starting with getting the drums sounding good, but with a proper production and a great song like this I first listen to it a lot. When I have a connection with a song I tend to do less nitpicking and mix more from an overall picture. I do listen in solo mode at times, of course, but you have to be careful not to go down a rabbit hole, as your mix may start to sound disjointed. Everything has to fit, so I mix a lot with everything in.

“Like most people I generally start with the

rhythm section, but in the case of Downtown I began the mix by focusing on Ben’s vocals. The drums were already in a good place, though perhaps not quite hitting as hard as they are now, so I focused on getting the vocals to be in charge, and then built everything around them as support.

After Ben’s vocals I worked on Eric’s vocals in the chorus, and then the rest of the vocals, so they told me the story of what needed to happen between each section. After that I worked on the horns, followed by the drums and bass, really making sure the groove was solid; and then I worked the other instruments in.”

Somehow, Castelli managed to make everything work together. Over the course of five and a half minutes, the song takes you on a journey that is partly a throwback, definitely inventive, and, most importantly, fun. Pop songs aren’t typically this diverse, and Castelli’s mix makes those transitions effortless so you can just get down without a thought for the business of how it all came together.

LEAD VOCALS Castelli: “Ben’s main vocal track sounded really good, and didn’t need too much work. I have five plug-ins on it. The FabFilter Pro-Q2 is a high-pass at 100Hz and also takes out some 700-900Hz — a frequency range that I tend to dislike in vocals. The UAD API Vision Channel Strip adds a nice aggressive sheen.

You can make it sound hi-fi if you want, but it’s best suited to that solid-state aggression with a rock edge that I wanted here. I boosted high end and took out some 240Hz with it, and also used the compressor, set to a ratio of 3:1.

The UAD Pultec Pro is boosting 200Hz, adding some broad warmth, plus I also had the FabFilter Pro-DS. Ben’s vocal has no reverb. They don’t like reverb on his voice; I only used it on him in one song.”

BOOST YOUR IMAGE CONSOLE VS THE BOX Castelli: “When I was nearly finished with the mix, I went to Ben and Ryan’s studio in Seattle to work the three rappers they’d recorded into the session. They asked me to put the mix through a console, so I rented EastWest Studio B to use the vintage Neve for a more classic 1970s sound.

It didn’t work, it didn’t sound as good as my in-the-box mix that had gone through my Rascal Two-V and UBK Fatso. We spent $2000 for a day, and it didn’t sound any better!

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“I mixed a song for Robbie Robertson on his Neve at the Village in LA, as well as in the box, and via a Dangerous summing mixer. In that case we also ended up using the in-the-box mix, because it sounded better. The Neve and summing rig made it sound mushy; the mix lost its edge.

“That said, I recently mixed a couple of songs at a studio in Burbank on Chris Lord-Alge’s old SSL. I don’t think I could have made it sound the same in-the-box. I also went to Blackbird in Nashville a month ago and summed a mix through a console, and again it sounded great. I’m open to either way of working. In-the-box is definitely easier for recalls, but as far as the sound is concerned, you can mix in-the-box or on a console. It depends more on the room, the monitors, and most of all, the song.”

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ERIC NALLY VOCALS

HORNS

Castelli: “Eric’s main vocal track has eight plug-ins on the inserts and a send; though all the plug-ins are making subtle changes. I started his vocal chain by hitting the Waves API 2500 compressor pretty hard, then I boosted 100Hz and took out 500Hz on the Waves API 550B. He has a quiet voice, so I needed to give him lots of body. As with Ben’s voice I used the FabFilter Pro-Q2 for its hi-pass filter and ducking around 700-900Hz. The FabFilter Pro-Mb is controlling some of the 800-2000Hz range that comes from the mic. The Softube TubeTech PE1C EQ is attenuating some 10kHz, and the Plugin Alliance Maag is boosting 160Hz, 650Hz and 2.5kHz. Finally, the Softube TubeTech CL1B is doing some mild compression. The send is a UAD EMT140 plate, set to a short, bright reverb, with a long pre-delay, to give his vocal a short but big glow, so he sounded huge. His vocals are also parallel compressed using a Universal Audio LA 610 and a Bomb Factory BF76.”

Castelli: “The horns are a big part of the song. You don’t get many horns in pop music these days, so I wanted to make sure they sounded special. I wanted a grimier, darker horn sound, rather than the ’70s funk sound the track already had. I opted for a dark tape machine vibe, with very little compression. It was more about filtering the top end and finding resonant peaks so it cut through the track without sounding too fancy. The horn tracks are all sent to a bus on which I placed the UAD Studer A800 tape machine emulation, the Waves SSL E-Channel to try to make it sound more classic, the Softube TubeTech ME1B EQ dipping around 2kHz and boosting at 200Hz and 5kHz, the Pro-MB, and the ProQ2 which is a hi-pass. Mostly it’s about the A800, which is set to 15ips, aiming to make it sound darker.”

EXTRA VOCALS

DRUMS & BASS

Castelli: “Towards the bottom of the session are the rest of the vocals, including the big choir with about 60 voices compiled in stems and the three rappers. All of them have no plug-ins at all. They were more about balances and rides. Many engineers feel they have to do something on each track. I may have used a lot of plug-ins on Ben and Eric’s vocal tracks, but in general I subscribe to Tony’s school of thought, which is that you don’t have to do much, and you only do things when really necessary. If it’s already good, don’t touch it!”

Castelli: “The drum tracks fall roughly into two categories. The more obviously programmed drums are sent to a bus with the Studer A800 and parallel compressed using a dbx160. The rest of the drum tracks comprise drum samples that aim to sound live. The drums already sounded great, so I just used EQ to make them sound a bit harder and added a sample to reinforce the kick sound.

The bass also didn’t need much. I put on a bass amp plug-in and Culture Vulture parallel compression to make it gel with the drums. It was all about the glue.”

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STUDIO FOCUS:

MERLOC STUDIOS As of September, Canberra will have a new studio on the scene. ACT-based producer Sam King will be opening his new recording space Merloc Studios. King is no stranger to the local music scene in Canberra, over the last 12 years he’s toured and performed with The Ellis Collective, Mr Fibby and Burrows.

Recording has always been a passion too, starting off as a wet-behind-the-ears 14-year old with a four-track cassette recorder. Up until now, King had been honing his craft in his home studio, producing records for artists such as Owen Campbell, Julia & The Deep Sea Sirens and Cracked Actor. With enough jobs under his belt to warrant an expansion, King set about constructing Merloc Studios, which has taken two years to finish. But worth it. “One of my main objectives in the design of the Merloc Studios was for it to be the kind of space that inspired you to want to play music,” said King. “I wanted it to have all the industry standard functionality and also be an aesthetically pleasing space.” And the nation’s capital will be the better for it. www.merlocstudios.com

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REVIEW

ZOOM F8

Multi-track Field Recorder Zoom isn’t afraid to innovate, and while the F8 might not usher in a new field recorder category like the H4, it’s certainly dreamt up a dramatically new pricepoint for it.

Review: Greg Simmons FRONT PANEL — As with all portable field recorders designed for overthe-shoulder operation, panel space is at a premium and the F8 wastes none of it. It’s a tight fit and therefore a little fiddly, but Zoom has managed to keep it acceptably fiddly. A clearly numbered record-enable button, rotary gain control, six-segment LED level meter and PFL button are all neatly partitioned for each channel and accessible enough. Just to the right of the power switch is the built-in slate microphone. All of the push buttons and switches have a firm and positive click, and none of the knobs feel loose or wobbly. Overall, very reassuring.

NEED TO KNOW

SCREEN Xxxx Xxxx — To give you clear-as-day operation outdoors, the full-colour backlit LCD screen can be switched to an ultra-bright monochrome mode.

Immediately to the right of the display is a toggle switch for recording slate tones or voice messages, a rotary control and pushbutton for menu navigation, and the headphone volume control.

PRICE Expect to pay $1699 CONTACT Dynamic Music: (02) 9939 1299 or info@dynamicmusic.com.au

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PROS Affordable Interface doesn’t require manual to operate Excellent form factor with confidence-inspiring build Good sound Two modes of operation — global & channel specific

CONS Lack of pads on microphone preamps Headphone amplifier a let down Tactile mixing via app only

SUMMARY Game changer. While on-the-go mixing isn’t its forte, Zoom has packed enough into the F8 package to once again change field recorder price-to-performance expectations dramatically.

Zoom blipped onto my radar in the early ’90s with its nifty little 9002 guitar effects processor; a device small enough to hook onto a guitar strap with a remote control that mounted on the guitar body. Although I admired the innovative thinking, it was of no use to me because I’m not a guitarist. Zoom didn’t have much to interest me as a sound engineer until it released the H4 Handy Recorder in 2006 — its first venture into the recording market. With two built-in microphones and two XLR mic inputs in a battery-powered handheld recording device, the H4 ticked a lot of boxes in the price-versus-performance checklist.

I’ve met many users who swear by their H4s, and just as many who swear at them. The design was not without its flaws: the user-interface was clumsy, the LCD screen was small and dim, and the noise of the external mic preamps rendered them useless. Then there was the bizarre stereo technique used for the built-in microphones; supposedly a coincident pair of cardioids at 90 degrees, but with the diaphragms spaced a small distance apart and facing towards each other in what I referred to as the ‘cross-eyed’ arrangement. The H4’s manual went to great lengths to incorrectly explain why this was a good technique — something I’m sure Zoom copped a lot of flak about.

The H4’s successor, the H4n, addressed all of those issues. The user interface was improved with important buttons relocated and a larger, brighter screen. The external mic preamps were also improved, and the cross-eyed mic technique was replaced with a truly coincident pair. The H4n’s user manual went to great lengths to correctly explain why this was a good technique, and in the process criticised the H4’s cross-eyed technique… while carefully avoiding self-incrimination! Not only did Zoom get the microphone technique right, it developed a clever method to change the subtended angle from 90° to 120° by simply rotating the individual microphones. I still consider it ingenious.

The point of starting this review by discussing much older products is because it says important things about the people at Zoom. They’re not afraid to innovate, they take user-feedback seriously and they are fast learners. Comparing the original H4 with the more recent and similarly form-factored H6 shows just how willing Zoom is to respond, improve and innovate. Which brings us to the F8.

F FOR FIELD

The F8 is an eight-input, 10-track field recorder designed for over-the-shoulder use in location sound recording for film and video. F for ‘field’ as opposed to H for ‘handheld’, I assume. It has eight mic/line inputs that can be recorded to eight separate tracks, and an internal eight-into-two mixer that allows a stereo mix to be recorded at the same time (hence, 10 tracks). The stereo mix can also be routed to three different outputs — headphones, main, and a sub output for feeding a DSLR or the like. Each features an individual output level control and peak limiter.

The F8 can record 16- or 24-bit .wav files at sampling rates from 44.1k to 192k, and mp3

files from 128kbps to 320kbps. The F8 has two SD card slots and is capable of recording to both cards simultaneously. It can also record different things onto each card in real time – for example, multitrack files on one card and a stereo mix for the dailies/rushes on the other. It also features a dual record option when using four or less inputs that allows each input to be recorded to two separate tracks, one at a lower level as a safety in case of excessive levels.

The whole thing weighs under 1kg without batteries. It has comprehensive timecode capabilities, can be remotely controlled by iOS devices, and can also be used as an audio interface with a Mac, PC or iPad via its USB port.

Considering all that the F8 packs into its small size and price, I was expecting its sound quality to be the disappointment. The truth is, however, that I honestly cannot be critical of the F8’s sound quality

ZOOMING IN & OUT

The F8’s preamps are padless but offer up to 65dB of gain on a single rotary control (+10dB to +75dB on microphone inputs, and -10dB to +55dB on line inputs). The minimum microphone gain of +10dB was not a problem during testing, but experience with other padless preamps has shown that a minimum gain of +10dB can still be too much. It’d be wise to carry a couple of in-line pads if you plan to use microphones with high sensitivity on loud sound sources.

The combo inputs automatically switch to line when a 6.35mm jack is inserted. Phantom power is switchable on each individual input, with a global option of using either +48V or +24V; running the latter, when possible, conserves battery power. In a nod to the consumer market, each input also offers Plug-In Power. Also known as PIP, it’s commonly provided by smart phones and similar devices to power electret condenser microphones with unbalanced outputs.

Unfortunately there was very little information about its implementation in the manual.

Each input has a high-pass filter switchable from 80Hz to 240Hz in 10Hz steps. I can find no mention of the filter’s slope in the manual or otherwise, but its wide range of cut-off frequencies allows it to do the job for those times when you must use a HPF during capture rather than waiting until post. There’s also a comprehensive limiter on each input with options of hard or soft knee operation, adjustable threshold (-2dBFS down to -16dBFS), adjustable attack times (1ms to 4ms) and adjustable release times (1ms to 500ms). The ratio is fixed at 20:1. It is, after all, a limiter.

A welcome feature that is missing on many contemporary devices is the ability to invert the polarity — incorrectly labelled ‘phase invert’ on the F8 but you get the idea. In addition, each input has a built-in delay adjustable from 0ms to 30ms in 0.1ms steps (perfect for synchronising different input sources), stereo linking capabilities between adjacent channels, optional MS decoding on record or monitoring, and a very useful PFL button that offers a choice of PFL or Solo behaviour.

How is it possible to offer so many input channel features at such a small size and low cost? The first stage of the analogue input path is a AT 57

THREE-WAY POWER — The F8 can be powered from three different sources: the supplied AC adaptor, an external DC input on a four-pin Hirose connector that can accept 9-16V, and the removable battery cartridge that holds eight AA cells — alkaline, nickel-metal hydride or lithium batteries can all be used. According to the manual, battery life varies from over 12 hours to just over one hour, depending on the type of battery (alkaline, nickel-metal hydride or lithium), number of channels/tracks, sampling rate, intensity of meters and use of phantom power.

A user-defined voltage level determines when the F8 will automatically switch from external DC to batteries, in the event of insufficient or lost external power.

Numerous power management options allow the battery life to be extended by turning off unwanted circuitry. Purchasing more cartridges and stuffing them with batteries would allow very quick battery changeover in the field. I would not be surprised to see a lithium ion or similar rechargeable battery pack and charger on offer from Zoom in the near future.

CONSTRUCTION — My reasoning for opening the F8, other than having a reviewer’s license to be nosy, was to check the construction of the chassis — an important consideration for a field recorder. At first, I removed the four large flat-blade screws on the top panel, only to find they exist purely for mounting the supplied camera-mounting bracket. Cleverly, removing these screws doesn’t open the chassis, so there’s no chance of damaging the internals while accessorising in the field. The top, bottom and side panels appear to be made from 3mm brushed and anodised aluminium, which form a rugged sandwich and provide strength where force is applied to connectors. The custom-shaped front and rear panels are both made of plastic. It’s well done, the combination of plastic and aluminium has strength where it’s needed while keeping the overall weight down.

GOT DATA COVERED — The two SD card slots have magnetically-latching protective covers, and the mini-USB port is far more reliable than the ridiculous micro-USB ports appearing on many portable devices these days.

combined microphone preamp and A/D converter.

From there on, all processing takes place in the digital domain. Some readers might assume it’s pointless having the limiter after the A/D converter, but through clever use of digital gain staging and by making the limiter work directly on the preamp/ AD converter (rather than its own variable gain cell) it’s not a problem. This technique has been used by Nagra and others in recent years, and requires a shift away from traditional analogue thinking to fully appreciate.

FINDING FINGER LOCATIONS

One of my major bugbears with location recorders is the user-interface; the likelihood of fiddliness increases with inverse proportion to panel space. There is just enough front panel room to access all of the controls with reasonable certainty although, as with any small piece of technology, keep an eye on what you’re doing if you have particularly wide fingers!

The F8 relies on a simple method of navigating through its numerous menus and functions based on a Menu button and the Select Encoder — a AT 58

rotary dial with a built-in press button. Pressing the Menu button opens the menu system. Turning and pressing the Select Encoder does the rest, assisted occasionally by one or two of the transport control buttons.

The menu system accesses all the features of each input channel in a feature-by-feature manner.

For example, changing the settings of the HPF for one particular channel via the menu system means selecting HPFs, selecting the channel, making the adjustments, and then pressing the Menu button to reverse out and move on. This back-and-forth menu navigation gets a bit tedious when you’re trying to set up all the input options for one channel in particular.

Thankfully, the F8 offers a much quicker method: pressing the PFL button of any armed channel brings up all of its settings on one screen, including input gain in dB, phantom power, high-pass filter, limiter, polarity invert, metering, monitor mix fader and pan. It’s a much faster way to quickly tweak the parameters of an individual channel. My favourite method of working was a hybrid of both — use the menu system for

setting global and system parameters, and the PFL function for controlling channels.

When not accessing the menu system, rotating the Select Encoder provides access to the internal eight-into-two mixer along with four customisable metering views. The meter ballistics include Peak, Peak + VU or VU only, with adjustable peak hold times.

It’s a testament to Zoom’s interface design that I didn’t open the manual until after making many test recordings with the F8. In fact, the only reason I consulted the manual was to double-check facts while writing this review. Oh, and also to find out how to make the Bluetooth connection to an iOS device. Speaking of which…

IN THE iOS MIX

To control the F8 remotely via an iOS device requires downloading the Bluetooth driver from Zoom’s website, copying it onto an SD card, inserting it into the F8 and delving into the menu system. Of course, you must also install the appropriate app in your iOS device.

The iOS app provides a much more spacious

LOCKING IN — The rear panel has two recessed BNC sockets for timecode in and out and a multipin socket for connecting any of the optional microphones that Zoom made for the H6 (any microphones connected here replace channels one and two). An added touch in the lower left corner of the rear panel is a Kensington Security Slot – bring your own cable and lock, as always.

HEAD SCREWED ON — The 6.35mm TRS headphone socket (good!) and minijack Sub output (for connecting to prosumer cameras) both use metal threads and nuts. This is a major improvement over the minijack sockets found on Zoom’s series of ‘Handy’ recorders (H4 to H6). In my experience with many H4s, H4ns and H6s, that socket is the first thing to fail due to plastic enclosures the lack of sufficient mounting strength. I can’t imagine that happening with the F8.

and faster interface than the F8’s front panel but follows the same menu system so there’s nothing new to learn. The downside of that approach is that, aside from multi-touch mixing, it fails to take full advantage of efficiencies offered by the iPad interface. For example, touching a menu item to reveal another menu with two options that could’ve simply been presented as a dropdown menu on the first touch. Hopefully later versions of the app will be more refined.

One of the knocks on the F8 is its limited hardware mixing control. Which for some applications, is a must. The app at least provides a means of multi-channel mixing, while letting the F8 do what it does well; pack in loads of channels and tech at a ridiculous price point.

FOUND SOUND QUALITY

The problem with reviewing the sound of audio gear these days is that nothing sounds ‘bad’ per se; what’s ‘bad’ today sounds amazing compared to what was ‘bad’ a decade ago. Good quality mic preamp and converter chips are highly affordable to manufacturers, and are therefore making the tonal differences between the cheapest and most expensive gear harder to discriminate. Judging differences requires a wider variety of sound sources to highlight them, and considerably more listening time. To that end I gave the F8 quite a hammering during the review period, making numerous recordings including direct-to-stereo piano, multitrack jazz, outdoor field recordings and dialogue.

In some cases I used Y-splits and made simultaneous level-matched recordings to other machines to make valid comparisons. Those machines included a Nagra 7, a Sound Devices 702 and an Apogee Quartet. It’s worth bearing in mind that the Nagra and the Sound Devices are both two-channel machines costing considerably more than the eight-channel F8, while the Apogee

is quite a different device but worthwhile for the sonic comparison.

Considering all that the F8 packs into its small size and price, and the fact that I could find nothing to criticise in terms of features, build quality or ease-of-use, I was expecting its sound quality to be the disappointment. It would’ve justified the old adage, ‘you get what you pay for’ and made this review a whole lot easier. The truth is, however, that I honestly cannot be critical of the F8’s sound quality.

Subjectively, it sounds marginally brighter than the more expensive Nagra 7 and Sound Devices 702, but that brightness manifests as a subtle sheen that provides an enhanced sense of clarity without any suggestion of harshness or brittleness. I did not find it tiring or fatiguing.

At the same time, the F8 does not have quite the same fullness in the lower midrange as either of the other machines. The end result is an ever-soslightly ‘lighter’ sonic presentation. I must stress that these differences are very subtle and should be considered more as a characteristic tonality of the F8 rather than a criticism.

F8 SOUNDS You can judge for yourself by downloading the comparison sound files I have provided with this review at soundcloud.com/audiotechnology.

I should add that these sound judgements were all made after loading the files into a DAW and monitoring through decent speakers and headphones. The built-in headphone amplifiers on most location recorders should never be trusted for this kind of comparison, and especially not the F8’s headphone amplifier. If the device has any weaknesses, the headphone amplifier is it. Listening to F8 recordings through

SOLID OUTPUT — A pair of TA3 sub-miniature XLRs (TA3 to male XLR adaptor cables included) provide the main stereo outputs.

It’s worth noting that all of the combo sockets are screwed to the aluminium side panels, relieving the solder joints from any excessive force applied to the XLR or jack – a common occurrence in the field.

its internal headphone amplifier, and then through a decent headphone amplifier, was quite a surprise. I guess headphone amplifier chips have not yet made the same advances as mic preamp and A/D chips.

CONCLUSION

Zoom’s F8 is a feature-rich multitrack field recorder with an internal complexity that has been rendered simple through clever interface design.

It is surprisingly straightforward and free of any operational quirks and gotchas, and its clever features reflect a design culture that has spent many hours listening to end users. It’s ruggedly built, weighs around a kilo with batteries installed, feels solid and confident, looks good and sounds good. Add an iPad for remote control and it gets even better.

All things considered, it’s fair to say that the Zoom F8 is a game changer. How it stacks up sonically against competing devices is a matter of opinion, but anything costing more than twice the price of the F8 is going to be a hard sell from now on.

I fully expect to see the F8 in the kit of every serious location sound engineer in the near future — either as their primary recording machine or as a cost-effective extension or backup.

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REVIEW

Bose F1 Flexible Array It looks like a point-source box, but Boseâ&#x20AC;&#x2122;s F1 portable system has an array of unique features.

NEED TO KNOW

Review: Mark Woods

PRICE $1699/piece CONTACT Bose: 1800 173 371 or f1.bose.com.au

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PROS Line array clarity Flexible vertical throw Portable design Built-in stand

CONS Too smooth for hard rock

SUMMARY Bose has taken a typical point-source-looking cabinet and filled it with a mini-line array. With a horizontal spread of 100-degrees, variable array angles, 1000W on tap, and a companion sub; this portable package will fill rooms you never could with a plastic box.

Bose does things differently. There’s lateral thinking behind many of the products in its large range of consumer and industrial products.

Audio systems for live sound is only a part of what it does and Bose doesn’t release new offerings very often. When it does, Bose always seems to have a different way of finding new solutions to old problems. The well-liked 800 series from the late ’70s had a processor and 8 x 4-inch speakers. Light and easy to move around, they still sound okay today. The L1 Series uses line array principles in a portable system. It’s been around for over 10 years and still finds favour with performers who enjoy its ability to cover both the stage and audience with clear, even sound. Not surprisingly the new Bose F1 system is different again. A hybrid approach, it uses elements of conventional powered speakers combined with line array techniques to create a specialist FOH system with a wide throw and enough power to fill a small- to medium-sized venue. The ‘surprise and delight’ feature is its ability to control its vertical coverage angle.

CLICKING INTO GEAR

Much of Bose’s innovation is housed in the F1 system’s Model 812 full-range speaker. It’s a similar size and weight (20kg) to other powered speakers but the design is very clean with excellent recessed handles on the top and rear. Made from some tough composite plastic — in regulation charcoal-grey — the finish is subtly textured and patterned, but it’s the front of the speaker that makes you look twice.

Looking past the perforated steel grille, instead of the usual horn/woofer configuration there’s a vertical array of 8 x 2.25-inch mid/high drivers in there. Closer inspection reveals a 12-inch woofer mounted further back in the box. The array inside the laddered centre section and prominent centred Bose logo ensures a distinctive corporate look.

Inside the cabinet there’s processing, protection and 1000W of power. Connections on the rear are all familiar with two inputs; one is a mic/line XLR, the other a choice of stereo 6.5mm or RCA sockets.

The 812 works on its own as a full-range speaker; it’s pole-mountable and provides good low frequency response (-3dB @ 52Hz) that’s more than adequate for speech and medium level music applications. But for live music or DJs this system benefits more than most from being combined with its matching subwoofer. The F1 Subwoofer houses two 10-inch drivers and a 1000W amp in a cabinet designed to fit in a car. Like the 812, transport is made easier by good handles on the top and rear.

Construction is from wood with a composite plastic base and lid. It weighs in a little heavier than the 812 at 25kg but it’s still manageable for one person. The low-end frequency response is strong from around 40Hz, this allows the 12-inch woofer in the 812 to be crossed over at 100Hz so it can concentrate on the low-mids instead of trying to reach down too low.

A great feature is the built-in stand — ‘extension bracket’ in Bose-speak. Instead of the usual polemount on top of the sub, Bose has made a plastic frame that clips onto the rear of the sub cabinet for

transport. At the show it slots into the top of the sub to support the 812 mid/high speaker and before you know it you’ve got a time-aligned speaker stack standing two metres tall. Not only is it a unique look but it’s quite stable, the solid base is not as wide as tripod legs but it’s squarer and harder to trip over. You may never have to deal with speaker stands again and I bet you won’t miss them.

F1 HAMMERS THE BENDS

Line arrays produce a wedge of sound that is wide horizontally but narrow vertically. The size of the array determines its effectiveness at lower frequencies, so to be practical, portable line array systems only deliver the array benefits at mid/high frequencies. The subs are usually conventional designs and omni-directional. The F1 stack has a forward-pointing single 12-inch woofer above the sub so it’s a fairly normal, almost point source system up to the 600Hz crossover point where the mid/high array starts to do its thing. Dispersion is quoted as 100 degrees horizontal and 40 degrees vertical. Listening to the 812 up close, they sound good right on axis with a smooth quality that refuses to bite or feedback. The mid/high frequencies roll-off strongly above or below the central axis compared to point source designs, although the woofer has a more normal conical dispersion.

The wide, even coverage and long throw high-mids mean good intelligibility over a large area while the lack of harshness makes for a relatively easy listening experience

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The ‘surprise and delight’ feature is its ability to control its vertical coverage angle

Arrays are weird up close anyway as the sound needs some distance to develop. That combined with the super-narrow vertical angle rules these speakers out for stage monitor duties. Bose has thoughtfully not provided a monitor angle in the design, and the cabinet looks better for it. Using powered FOH speakers as floor monitors is always a compromise, but a common one, so there’s some lack of flexibility here compared to regular powered box designs.

These speakers do their best work as FOH speakers in small-to-medium rooms with low-tomedium volume acts. Arrays focus the sound and reduce reverberation by not throwing sound all over the room. The most noticeable benefit of this is mid-range clarity, right where the vocals are.

GOT YOU COVERED

Sutton’s House of Music in Ballarat is a textbook case of a room that could benefit from a line array system and a good test for the F1. The 19th century building was a perfect piano sales floor; long room, high ceilings and naturally live. It’s a goodsounding venue but reverberant, and crucially, the bands play across the width of the room, requiring the speakers to cover a wide area.

People eat and talk, others listen or dance to the solos/duos/combos and other non-rock acts that play there. The sound needs to be clear but not loud or piercing, with intelligible vocals throughout the room and enough low end power to get people moving.

We set it up for a night featuring regular act B3 Breakout; a three-piece ensemble featuring a Hammond B3, drums, and guitar and vocals. The F1 system was very quick to set up and sounded ready to go straight out of the box. The room had a boom that required some low frequency EQ but the mids and highs sat close to flat. The noticeable impression from the sound check was the F1’s ability to fill the room. The subs were strong and could have filled a much bigger space, but the mids is where you could hear the difference compared to horn-loaded boxes. The vocals surround you rather than come at you directly from the speaker. They’re not loud up close, but step back a little and it’s all there. The high frequency response is adequate for live sound, although somewhat lacking in transient AT 62

detail. They don’t want to feed back, they seem to find their own level and they’re very even around the room.

The F1 system’s horizontal coverage of 100 degrees was just wide enough. The high/mid clarity falls off sharply at the edges, but it is a wide room that normally takes four speakers to get adequate coverage. I found them easy to mix on and throughout the night, there was plenty of unprompted confirmation from customers and staff that the F1 was a superior solution.

Narrow vertical angle usually helps live sound but sometimes the audience is positioned above or below the speakers. Horn-loaded boxes often throw high or low enough (or offer different angles for the pole mount) but to get similar benefits from an array it needs to be focused. The F1 812 addresses this by allowing the edges of the top and bottom sections of the front grille to be pushed in and out. This angles three of the eight little speakers up, or down in the case of the lower section. They snap into position with magnets and the internal processor makes some compensatory changes to the EQ. Between the top and bottom angled sections you can create four distinct vertical patterns that can be used in situations where the

speakers are above and/or below the audience. It’s a neat approach and necessary too as they sound pretty dull if you’re not in line.

WIDE APPEAL

Another place arrays work well is outdoors and the F1 system is powerful enough to cover mediumsized events. The wide, even coverage and long throw high-mids mean good intelligibility over a large area while the lack of harshness makes for a relatively easy listening experience. This is usually a good thing but it’s also the limiting factor with portable line arrays. If the going gets loud it can expose the relative lack of bite and body on offer.

Point source, horn-loaded speakers are beamy and squawky but they will cut through a loud band.

The Bose F1 System will appeal to venues, bands, DJs and groovy bars but it would also be ideal for corporate presentations or speeches. It’s fairly easy to transport if someone in the band has a wagon and could provide sound for over 500 people in the right situation. The price is pretty on par with similarly powered portable systems, but it’s got some unique features and the high/mid array will allow it to out-perform point source systems in many audio environments.

Studio B

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Studio C

Studio D

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www.studio52.com.au AT 63

REVIEW

DI SHOWDOWN We pit one of the earliest passive DIs ever made against the latest all-tube, standalone tone shaper.

NEED TO KNOW

Review: Greg Walker

PRICE $2066 CONTACT Mixmasters: (08) 8278 8506 or sales@mixmasters.com.au

PROS Active Baxandall EQ Standalone, no extra preamp required Separate Main & Aux output controls

CONS Expensive, but can double as a preamp

SUMMARY This do-it-all tube DI is everything you want in a DI. It can even stand in for a mic preamp, with Tonecraft’s impedance adaptor in front.

Tonecraft 363 Tube DI vs ACME Motown WB-3 DI

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

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PROS Recreated Motown-era transformers Choice of preamp pairing

CONS Up there for a passive DI

SUMMARY For true recreation of that Motown sound, Acme has done the hard work of recreating the original transformers. But you’re going to need a tube pre from that era for extra authentic vintage vibes.

NEED TO KNOW

PRICE $732

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Tonecraft is a relative newcomer and its debut product is the well thought out and very comprehensive Tube Direct Box. This is quite a hefty and pro-looking black desktop (or floor sitting) device that takes up about half of a two RU space and is quite deep to accommodate the hand-wired component layout. The build quality is excellent, with aesthetics to match.

The unit utilises chicken head knobs and stainless steel switches that, along with the thick steel plating, give the unit a real presence. Powering up the unit reveals a lovely light green backlit window with clear labelling of the main controls — quite a funky and unique look that works a treat. The key features here are the 6SL7 and two 6SN7 tubes, the two-band Baxandall EQ and the provision of dual outputs. Main and auxiliary outputs are on XLR and TRS respectively and also feature independent continuously variable gain and ground lift (on switches round the back). The front panel features large continuously variable volume control, bass and treble controls, EQ bypass, TRS input and an additional ‘thru’ TRS socket. The Baxandall equaliser is a classic design that utilises an active tone stacking circuit, and here there is a generous +/- 20dB of gain enabling some quite dramatic tonal shaping.

The dual output encourages studio users to work some parallel

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TONECRAFT 363 TUBE DIRECT BOX

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For most of us, the direct injection box sits in a somewhat unfancied corner of the audio universe; well beneath the level of desirable microphones and outboard devices but just above the purely functional attraction of patch bays and cabling. I suspect we should all care about DIs a lot more than we do, especially when you consider how many direct injection signals make up the very bone structure of our bottom end in mix after mix courtesy of the electric bass guitar, not to mention the capture of keyboards and other 1/4-inch jack-equipped, high impedance sources. It would be interesting to know what devices the average AT reader uses as their main DI solution. If it’s something cheap and cheery, you may find you pay a high price come mixdown. The Tonecraft 363 Tube Direct Box and the Acme Motown WB-3 DI both answer the direct injection question in different ways, and both these answers should give readers food for thought — if they are in the habit of plugging straight into their digital interfaces.

WITH MACKIE

INJECTION CORRECTION

iPads sold separately

It’s not every day a brand new DI turns up on your doorstep, so it was quite a surprise when the good folks at Mixmasters sent me a box containing not one but two very different DIs to review. Little did I know this was to be the beginning of an epic battle: tubes versus transformers, gain versus attenuation, feature rich options versus minimalist simplicity, David versus Goliath if you will. Could there be one clear winner?

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processing into the DI chain with compression being the obvious option. In a live setting the separate outputs can be sent to amplifiers and FOH. In this regard the separate gain controls for these channels mean greater flexibility in the gain structure, a nice touch from Tonecraft.

BASS IN YOUR FACE

Upon firing up the 363 Tube Direct Box for the first time it was immediately clear this was a quality piece of gear. Bass signals were clear and round with that unmistakeable vividness valve gear can impart on a signal. There’s tons of gain on hand and then of course there’s the EQ. It’s hard not to immediately reach out and start tweaking and the results are definitely very gratifying. Treble tones are sweet and defined, while not being too strident. While increasing bottom end does marvellous things to the scale of a bass sound.

Not to be discounted is the idea of going the other way and pulling out some top and bottom leaving a distinctive midrange tone. Either way there’s a ton of useable tone on offer and experience will tell you when to tinker or indeed when to bypass the EQ (as I did on a Perch Creek session) where a bass tone is sitting perfectly right off the bat.

Keyboard tones are equally at home running through the Tonecraft. My Korg MonoPoly sounded super fat through this DI and adding a bit of top and bottom gave the sounds new tonal dimensions. Similarly, DI’d electrics worked really well here, with great clarity and weight making this another good creative option. Interestingly, the folks at Tonecraft have developed the Boostmate adapter to allow the 363 to function as a mic preamp (predominately for dynamic mics), and this further extends the capabilities of what is a very sweet piece of kit.

ACME MOTOWN WB-3 DI

Where Tonecraft has gone for a comprehensive feature list, Acme has come in with a pocket rocket that is all about vintage sonic vibe. Not much larger than a standard guitar effects pedal, the Motown WB-3 comes in an army issue khaki gloss finish and sports just the one brown AT 66

chicken-head knob for level attenuation. Next to this a switch selects the use of this continuously variable attenuation or bypasses it to run the signal at full gain. Two 1/4-inch jack inputs complete the top of the unit’s controls while round the side are an XLR output and ground lift switch. Being a passive unit, the Motown WB-3 must use another source for downstream amplification so the unit is typically patched into a microphone preamp to generate proper recording levels. This makes the Acme a very different beast to the Tonecraft and in some ways quite hard to compare, as the tone of the Acme will always be influenced by the choice of preamp it passes through. The Acme design uses two painstakingly recreated custom wound transformers that are key to recreating the direct injection Motown bass sound of the ’60s and ’70s.

Where Tonecraft opted for the active approach, Acme has headed down the passive path keeping the design compact and simple.

TONE RIDER

My first recordings with the Acme WB-3 Motown DI were in tandem with the built-in mic preamp in my UA Apollo interface. I found the sound immediately engaging and enjoyed playing bass through this setup. When I switched to using the Chandler Redd.47 mic pre it did, however, take things to a whole new level. This set-up absolutely rocked and put me in a really great place for vibey bass performances. These takes needed no EQ and sat in the mix perfectly. Given that the Redd.47 is a very pricey mic pre this probably gave the Acme an unfair advantage over the Tonecraft, though I did also get great results passing the Motown DI through a much more affordable API 512b preamp. I did some experiments recording DI’d guitar through the Redd.47 and exploiting that unit’s propensity to break up under load. It was a tricky business tweaking the gain structure between the attenuation on the DI and I/O controls on the preamp, but there were some really interesting tones there that sat up beautifully in the mix and spread my guitar tone options quite a bit wider than usual.

SHOWDOWN LOWDOWN

In the end, choosing between these units really comes down to how you like to use DIs. The Tonecraft is a thing of beauty and looks built for the long haul. It’s a standalone solution for all sorts of direct recording situations — you put it on the floor for a session and it delivers lots of clean, powerful tone and your work is done. The Acme is more of a funky addition to an existing preamp or set of preamps, lending some of that vintage spice to whatever hits its circuitry and working some subtle magic in the gain structure there. Putting these two units head to head was a tricky test and it often came down to the particular song or part I was working on. I loved the Tonecraft’s grunt and EQ options for more straight ahead, cleaner tones. More often though, I chose the Acme for the way its sound matched my playing style and delivered an enhanced retro, funky kind of result.

The downside of the Acme however was that I lost a preamp channel which did balance out it’s relative affordability compared to the 636.

Having either one of these DIs in your studio will give you an expanded range of tonal choices and make you realise what you are missing out on when you plug straight into a recording interface’s preamp or an average preamp’s DI circuit. Once you get a taste for what a purpose-designed DI can do, it’ll be hard to go back to cutting that particular corner.

For the serious session bass player the Tonecraft is a great studio and live tool while the Acme will bring a big smile to many a funk, soul and rock bassist’s face; allowing an extra degree of play with any studio’s preamp collection.

www.gibsonami.com www.krksys.com AT 67

REVIEW

STUDIO ONE 3 PROFESSIONAL DAW Software

Presonus explores its creative side with the latest version of Studio One, and now you can too.

NEED TO KNOW

Review: Graeme Hague

PRICE $499.95 CONTACT Link Audio: info@linkaudio.com.au Or buy online at www.presonus.com

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PROS Arranger Track & Scratch Pad flexibility Send-less FX Chain routing Instrument chains High DPI GUI New Mai Tai synth covers plenty of bases

CONS Little light on in the drum & guitar department

SUMMARY Itâ&#x20AC;&#x2122;s all about flexibility of creativity in Version 3 of Studio One. You can change arrangements with impunity, break the chains off plug-in routing, layer instruments with ease, and recreate generations of sounds from a single synth.

The challenge for Presonus has been adding extra power, utilities and improvements, while staying away from that dreaded feature bloat

Considering how long some of the competition has been around, Studio One has become a very popular DAW in a relatively short period of time. Right from the start it’s been a rock-solid recording platform devoid of what Presonus calls ‘bloat’; a not-so-thinly disguised reference to the menu function baggage some DAWs carry, thanks to more than a decade of digital recording evolution. The challenge for Presonus over its subsequent upgrades has been adding extra power, utilities and improvements, while staying away from that dreaded feature bloat it disdains so much. With Version 3, is it still a lean, mean recording machine?

Well, it looks good. One criticism I’ve always had of Studio One was its rather bland GUI, particularly in the Mixer window. Some might call it ‘utilitarian’, others maybe ‘no frills’ — I’ll stick with bland. Thankfully, it’s had a makeover.

Version 3 offers a redesigned high-DPI display primarily to cater for its new touchscreen compatibility (yes, touchscreen) and the wide variety of devices where Studio One will be installed, ranging from compact notebooks to monster 5K monitors. Overall, the GUI is sharper, offers greater contrast and a slightly more 3D appearance to satisfy colour-and-movement junkies like me. Just to be clear, unless you’re applying the high DPI to large screens it’s not a massive difference, but I’m much happier.

Don’t freak out over the coloured channels, it’s a new option you can enable and I’ve gone overboard to show you. The improved GUI has brought variable-sized faders (previously it was simply long or short-throw faders), these colour options and a channel focus. It’s changed the Mixer from simply functional to a much more inspiring

GUI — yes, call me shallow and I don’t care. Some of you will agree that a Mixer View needs a bit of sparkle. Now Studio One has it.

Again, with the colour options and the improved DPI display you can channel your inner Mondrian with the Arrange View. An overall Hue setting in Options can shift the background colour from subtle to sickly, across the entire spectrum.

Aesthetically it doesn’t do much for me but I imagine it might be very useful for anyone suffering a visual impairment (such as colour blindness) that can be alleviated by removing or enhancing troublesome shades.

In the greater scheme of things relating to a DAW upgrade, it’s hard to get excited about an improved Browser, but Studio One’s browser has been given some treatment that streamlines file access and clip auditioning; and also includes thumbnails of effects and instruments.

SCRATCH THAT ITCH

Studio One’s new Arranger Track comprises labelled, coloured blocks along the timeline with

an Inspector list to the left. It’s a global track that allows whole sections of your song to be labelled — a verse or chorus, for example — then you can shift or copy all regions below it to your heart’s content simply by dragging around sections on the Arrange Track. It’s impressive just how seamlessly these sections can be moved around, including transplanting slices into otherwise whole audio files. Even without shuffling parts around, the large and colourful Arranger track is a boon for seeing where you are inside your project. It’s like markers that do more than just bump you around the timeline.

It’s rare to see completely new ideas in a DAW update, and Scratch Pad is one of those. Scratch Pads basically allow you to split the Arrange Window in two, with the Scratch Pad on the right, and main arrangement on the left. You can drag any of your Arrange track section to the Scratch Pad and perform edits in this alternative timeline (cue Doctor Who theme tune). The idea is to stuff around with your arrangements without any fear of trashing the original project, or confusing the AT 69

hell out of yourself. Say you want to work up an alternative intro, but you’ve always been afraid of messing with your main timeline, now you can drag it to the Scratch Pad, try out your idea and drag it back into your main arrangement if it makes the cut. Better yet, you can save multiple Scratch Pads in your project that contain different versions of your musical doodling. Even though you can only have one open at a time, it’s a much neater solution than sticking edits at a random spot down your timeline, or saving versions of projects you may never come back to.

FX CHAINS UNSHACKLED

Version 3 introduces just two new audio effects: a Leslie cabinet emulation called Rotor Audio, and Bit Crusher. Still, with Version 3’s new FX chain routing, which can now be manipulated beyond a linear channel strip order, every effect has been given a new lease on life. For example, a delayed signal can be split and fed into another plug-in or diverted around it to stay clean, or perhaps you’ll insert an EQ as a HPF sidechain on an effect that doesn’t have one. It allows the creation of unique effects, beyond the normal capabilities of your stand-alone plug-ins, without having to introduce load of sends and returns.

Presonus tells us that all its virtual instruments now benefit from a re-invented synthesis and instrument engine. I can believe it. New in the virtual instruments rack, Presonus has introduced Mai Tai, a polyphonic modelling synthesiser that gives you a great range of sounds from analogue Moog-ish leads and pads through to the latest AT 70

full-blown digital synthesis. As soon as you insert a Mai Tai instrument on a track, be prepared to lose several hours as you check out all the neat patches.

Its uncomplicated GUI encourages synth-illiterate users to dabble in creating their own.

Also kind of new, Studio One’s existing Presence sampler has been turbo-charged to a Presence XT version. Aside from extra control parameters you get additional XT Sound Sets plus three new libraries of loops and samples from Big Fish, Sample Magic and MVP — not to be sneezed at. What’s more, with Version 3’s introduction of multi-instruments, several instances of Mai Tai, Presence XT — or any other virtual instrument — can be chained together to create complex sounds. There’s a bit of a ‘suck it and see’ factor, because goodness knows what you’ll end up with when feeding one synth into another, but it’s great fun trying.

Exclusively in the MIDI department, the new Note FX function offers four modes for turning one-fingered piano pecking into virtuoso performances — Arppegiator, Repeater, Chord and Filter. Hardcore MIDI programmers will find the four GUIs a breeze, but novices without any knowledge of musical intervals might still struggle.

Nothing new about that when it comes to these sorts of MIDI effects. Figure it out and you’ll be able to play anything released by Seal in the last 15 years — including that awful song with his ex-wife singing harmonies. No wonder they split up.

On top of that, a Macro Control GUI has eight assignable controllers that will accept almost any plug-in parameter for fast access. And if

automation is your thing, Version 3 now has automation curves.

CONCLUSION

Studio One 3 has significantly enhanced the DAW’s capabilities as software for composing and arranging music, but it’s not just about the Arrange Track and Scratch Pad features. With the improved instrument engine, the new Mai Tai synthesiser and the multi-instrument layering (and Effects Chains) Studio One now provides an enormous palette of sounds and patches. Presonus gives you over 15GB of samples and users should rarely need to go searching beyond Studio One for that elusive patch — unless you’re talking about high quality drum samples or guitar amp modelling. Presonus includes its Impact library of drums, but it’s fair to say these fall short of third-party samplers like NI’s Studio Drummer. Studio One does have Ampire, which works well for quick and cool guitar sounds, but again it doesn’t quite measure up to dedicated guitar amplifier simulators. When it comes to keyboard and synthesiser sounds, and samples of acoustic and natural instruments, Studio One 3 has everything well covered.

It’s not the ‘complete’ DAW — will we ever see such a thing? — but Studio One 3 introduces some really handy new features and its bloat-free performance is still super reliable.

Toe dip: An Artist version of Studio One ships with Presonus interfaces, which can be upgraded for less than the full price of the Pro version.

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REVIEW

EXTREME ISOLATION EX-29 HEADPHONES Some 11 years ago, back in Issue 35, we first looked at a pair of Extreme Isolation headphones. So what’s changed in the world of ‘gun muff ’ cans? Not so much, as it happens. Extreme Isolation still has the market largely to itself: performance headphones with considerable acoustic isolation afforded by industrial strength ear cups.

The EX-29s are a more recent addition to the range and offer 29dB passive attenuation over the original Ex-25’s 25 decibels (the clue is in the name).

These ’phones are made in the US, and feature similar specs to the 25s: 40mm single drivers, 32Ω impedance, and a sensitivity of 114dB at 1kHz/1mW (which is more sensitive than the EX25s). They’re a comfortable headphone and they sound good.

I guess my assumption was that Extreme Isolation is happily sacrificing some sonic fidelity in the pursuit of maximum attenuation. And to a degree this is always going to be true — these aren’t designed as ‘close your eyes/stroke your chin’ reference headphones, they’re studio workhorses.

That said, they sound excellent. There’s a solid low-end, but not an emphasis that throws the tonal balance out, as you may have heard from bass-heavy DJ headphones. They’re not amazingly ‘transparent’ but there is real clarity. And for around $200 a throw, you can afford to have a handful of these in the live room and you won’t hear a word of complaint from anyone, I can guarantee it.

Closed back headphones are an essential in the studio. The advantage of using Extreme Isolation closed backs is you get an extreme lack of bleed into open mics, which can be a deal breaker for more delicate instrumentation.

To be clear — if you’re looking at it from an outside-in perspective — Extreme Isolation gets its attenuation from the gun muff-style design of the ear cups, not from any noise cancellation trickery.

Being ‘passive’, you don’t have to change batteries and I’d suggest you won’t feel any of the fatigue that’s possible with noise cancellation headphones AT 72

— which is constantly ‘sucking/blowing’ air into your ears to achieve its attenuation.

Live, I placed the EX29s on the head of a teenage drummer whose mum was concerned about preserving his hearing — which I was happy to encourage. This talented young chap was waiting on a pair of Ear Monitor Australia moulds, so in the meantime I thought I’d feed his monitor mix to the EX-29s. I think one good measure of just how good a pair of cans are is in how fastidious the musician gets about their monitor mix. And, sure enough, this drummer had me dialling him up a full-range mix including a touch of reverb — a far cry from the kick/bass/lead vocal mix he had in his wedge only hours earlier.

Extreme Isolation, the company, was born out of a drummer’s desire to safeguard his hearing over long periods of playing. This really is the EX29’s raison d’etre, and they perform admirably in this

regard — they’re comfortable and they feature industry-leading acoustic isolation. The slightly unexpected spinoff is that they’re well priced, they have a superior build quality (the jury is in after around 15 years in the marketplace; they’re pretty much bulletproof) and sound great.

If you’re a drummer, then I’m preaching to the converted — you’ll doubtlessly already know about the advantages of Extreme Isolation headphones and how they can assist in prolonging your playing career. Meanwhile, if you have a recording studio of any type and looking for a pair of live room headphones, then the EX-29s make a very compelling case. — Christopher Holder.

Price: $225 Mixmasters: (08) 8278 8506 or www.mixmasters.com.au

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AT 73

REVIEW

SHURE PG ALTA Microphones

Shure’s budget mics are all chips off the ol’ SM and Beta blocks.

But how close to the originals do they come?

NEED TO KNOW

Review: Mark Davie

PRICE PGADRUMKIT7: $925 PGA58: $99 PGA181: $175 PGA27: $349

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CONTACT Jands: (02) 9582 0909 or info@jands.com.au

PROS Quick release drum mic tensioners Great value PGA181 condenser option at dynamic price

CONS Handling noise Drum mounts not onesize-fits-all

SUMMARY Shure’s PG Alta series of microphones are good value, and sound close to their more established counterparts. If you’re in the market for a drum mic kit, but thought you only had the budget for a couple of mics, think again.

What is it with these budget mics and their ingenious mic clip tension clamps? It’s a revolution in anti-revolvers. Recently, I was impressed by Rode’s cylinder-clamping NT1 shockmount that tightened with the slightest grip of my thumb/pinkie combination. When I opened the case to peruse Shure’s budget PG Alta drum mic kit, the first thing I noticed about the tom mics was their bicycle quick release-style mechanism on the pivot joint. Surely that’s not new. But if it is, lets roll them out to every articulating joint on the market. They worked a treat; find position, lock in place — too easy. Compared to the slightly slippy traditional tensioner on Shure’s PGA27 condenser mic shockmount, the quick release versions were far superior.

Immediately I was getting a little excited about the build of these PG Alta mics; the second coming of the PG series which are priced at the bottom of Shure’s range under the SM series. The PG Alta range takes a few well-known Shure model numbers and adds another competitive option to the mix. So in the seven-piece drum mic kit, there’s the PGA52 kick mic and three PGA56 tom mics, derived from the Beta series; a PGA57 snare mic, drawn from — you guessed it — the SM57; and two SM-inspired PGA81 cardioid condensers for stereo overheads.

Rounding out the gaggle of mics I had on loan were the PGA27 large diaphragm condenser I mentioned earlier; the small diaphragm PGA 181 side address condenser — a tubbier version of the nifty Beta 181; and the PGA58 — no prizes there. It’s a bit of a mixed bag when it comes to how inspired these cut-down versions actually are. There has to be a little bit of money saved somewhere to warrant the budget figures. So where are the cuts?

PGA58 — SAME SAME BUT DIFFERENT

Probably the best case to investigate would be a comparison of 58s. The PGA58 sounds suspiciously like the SM version. Remarkably similar, in fact, with just a little less presence — like a good understudy. The switch on the PG Alta isn’t silent, but it’s not deafening either. I’d prefer it to have a more secure snap to it though. The main difference here is what makes the SM58 so special; its lack of handling noise. The PGA58’s motor assembly sits in a rubber-mounted cup, a similar design to a lot of other handheld dynamics. But arguably the best aspect of the SM58 is the elegant pneumatic shockmount Shure engineer Ernie Seeler devised for the Unidyne capsule. It basically drops handling noise altogether. The PGA was a pretty close approximation of the SM58’s sound, without the mechanical design bells and whistles.

PGA 181 — A really good candidate if you’ve been looking for a bit of a different flavour to a dynamic, but at a similar price point to one

PGA181 — UP IN YOUR GRILLE

I haven’t tried the Beta 181 yet, but the PGA181 makes me keen to. Using it on guitar amps — its natural home — it didn’t have that false, make my amp sound like a solid state-sound I find some condensers give me. It’s designed to be pressed right up at the grille, and it excels there. In that position, it can sound better than a mic worth 10 times as much that’s not designed for that purpose.

I suspect it has something to do with basket resonances. I typically use dynamics and ribbons in recording, and purely dynamics live, but I could see this changing a few of my preconceptions.

The PGA181 can handle 138dB SPL, requires about 18dB less gain than an SM57, to give you an idea. And because you’ve got it pressed right up against the grille, you don’t get nearly as much stage bleed as other condensers. It’s less bitey than

an SM57 — a smoother, rounded finish, and makes for a really good candidate if you’ve been looking for a bit of a different flavour to a dynamic, but at a similar price point to one.

PGA27 — SNUGLY IN ITS PLACE

The PGA27 large diaphragm condenser mic sits snugly in its little nest like a boiled egg in its plastic cup. Still, the low pass filter and -15dB pad on the back side of the mic are easy to get at. It’s a lovely looking mount that worked decently, though tapping the mic stand still made its way through.

Recording speech from a hand’s width away, the PGA27 didn’t have as much low end as other mics I put it up against. In fact, there was little difference between the normal and hi-passed takes at this distance. It’s a clear mic with a nice, slightly AT 75

exaggerated, high end presence that would make it suitable for vocal work. One letdown was that the frequency response changed more dramatically as I moved around the mic than other large diaphragm condensers. If your singer gets a bit off mic you might find some dynamic EQ coming in handy.

CLAMPING DOWN ON QUALITY

The build quality seems quite good across the board. The drum mics are robust; the grilles are tough, and unlike the free-floating PGA58 version, the diaphragm assemblies are all secured in place.

On the inside they’re mostly plastic, but precisionmachined plastic you could see lasting a long time.

The kit also comes with a zip up carry case, and enough clips and mic leads.

The rim-mounting system is simple to use, but a little limiting in a way that seems common for these systems. I have a custom Ayotte kit here with an isolation system that stretches most of the way around each tom. It only leaves one third of the rim exposed closest to the drummer; the most in-the-way position for a mic. The one-piece plastic part is designed to snugly hook over a standard rim, which didn’t quite work for my situation — something to keep in mind if you don’t want to use stands. The clamp angles away from the edge of the tom. So as you move the mic’s position away from AT 76

the drum head, it also moves closer to the rim — again, a little inflexible compared to other mounts, but a handy attachment for the right kit.

FULL KIT SOUND

I lined up the Shure PG kit against an Audix DP7 drum mic kit I use live. You get the same number of mics in both packs, but the Audix is about two and a half times the price. So keep that in mind.

The Audix D6 is a really simple-to-use kick drum mic, especially live. Its scooped sound doesn’t require a lot of tailoring. It has a more satisfying thud than the PGA52, but the 52 didn’t need a lot of help in the click department, which was good.

The PGA81 overheads were pretty well-balanced, had the tightness of focus you want from small diaphragm mics, and were a much lower output than the Audix small diaphragm condensers, which is perfect for overheads. They didn’t pick up a whole lot of low end, which was actually quite handy in live situations, where I’d usually engage a hi-pass anyway. The PGA56s aren’t overly detailed mics, but serviced the toms really well.

Compared to an SM57, the PGA57 snare mic was missing a little bit of the high-end snap, and consequently also a bit of the snare ribbon sound coming through from the underside. Nothing like the boosted 1-2kHz midrange of the Audix i5,

which gives it a pre-fabricated sound. It was really a case of getting brighter as you went from the PGA57, to the SM57, to the i5. So depending on where you sit with your appreciation of the SM57 as a snare mic, you could go either way with these two. I still preferred the 57 most of the time, but going through these mics again made me think the i5 was more useful than I’d been giving it credit.

It gave some nice snap, that brought out the snare ribbon sound in a very even manner, and brings the whole drum forward. They all had a similar level of bleed, but again the high end of the SM57 rendered any hi-hat bleed more ‘useful’, if you can call it that. It’s an industry standard for a reason.

While not as versatile as pulling together your favourite esoteric pieces from the cabinet, the PG Alta drum mic kit provided a really tight, one-stop drum sound. If my budget for a drum mic kit only extended this far, I’d rather have an entire mic kit that gave me the full picture than just a couple of mics to play with.

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Musos Corner 1 National Park St Newcastle West NSW 2302 PH: 02 4929 2829 www.musoscorner.com.au AT 77

REVIEW

ARTURIA KEYLAB 88 Controller Keyboard The Keylab has all the size, the keys, and the sounds you need. And importantly, the control to handle it.

The Keylab 88 has the same parameter control set as the 49- and 61-note variants. There’s the standard master volume, pitch and modulation wheels, and octave up and down buttons. From there you have two banks of 10 encoders, nine faders, 16 pads, transport control, 10 push buttons, and an LCD with parameter and value adjustment encoders.

Review: Mark Davie

NEED TO KNOW

Like all 88-note, weighted keyboards, the emphasis is on weight. Arturia’s Keylab 88 is not hugely different, but it is a bit lighter than most. That said, there’s a largesse to it that extends beyond the wooden end cheeks of its long metal chassis. Once you prise it free of its box, the Keylab can really stretch its legs out with a couple of attachable appendages. The first — a perspex-backed music stand — has an aluminium base to give it a firm, stable fit. Music sits on a rubberised base to stop books moving around willy nilly, though its grip impedes sliding individual sheets from side to side.

PRICE Expect to pay $1399 CONTACT CMI Music & Audio: (03) 9315 2244 or sales@cmi.com.au

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PROS Quality construction gives keybed solid feel Excellent integration for Arturia users Can sit a laptop on it

The second extension is a rubber-topped wing that adds some extra depth to the right side of the Keylab 88, augmenting the free space on deck into a paddock with perfect laptop-sized dimensions; my 15-inch MacBook Pro looked right at home up there.

It’s a great addition for any musicians keen on taking it onstage, especially since the Keylab 88 relies on a computer to generate sound.

From there, it’s just a matter of hooking up a USB cable, plugging in a sustain pedal and away you go. There’s provision for a power input, but being USB bus-powered, the Keylab doesn’t ship with one.

CONS Some GUI elements a bit small for live use

On the back, there’s also MIDI in and out; an expression pedal and auxiliary input, both of which can be assigned to any MIDI parameter; and a breath control input.

FEELING THE WEIGHT

Any decent controller is only as good as the feel of its keybed. Arturia has installed a Fatar TP/100LR unit; a decent keybed found in a lot of other name keyboards. It has a good, solid feel. If I was being nit-picky, the keys clunked to the bottom a little too readily. I like a key’s counterweight to hang in the balance a little more. That said, its action allows you to rip

SUMMARY Arturia’s Keylab 88 looks like a stage piano, and is built like one too. But with thousands of sounds bundled in, and Arturia’s fully-featured MIDI Control Centre, its synth credentials are what makes this less-than-humble controller a workstation replacement prospect.

across the keys with verve, which suits the supplied synth-heavy Analog Lab software to a ‘t’.

The surrounding unit has as much impact on the feel of the keyboard as the keys themselves.

And in this regard, the solid metal chassis of the Keylab 88 feels very rigid, with no warp when you really belt into it or apply force to the pads up top.

It’s a solid player.

The software supplied with the Keylab 88 goes a couple of steps further than the Analog Lab bundle, which in itself — at US$89 — is great value. As of this review, you get over 5820 sounds drawn from the 12 synths and keyboards in the Arturia V Collection. That’s almost 500 presets each, which amounts to plenty of variation.

Also included is the modelled Pianoteq 5 Stage, worth 99 Euros. Because Pianoteq’s instruments are modelled, not sampled, the packs are quite small in size and don’t tax your RAM like a sampled instrument does. When you first boot it up, you get to choose a style of instrument from three broad categories: traditional pianos, electronic piano sets, or vibes and marimbas. I’d recommend one of the latter two, seeing as you also get UVI workstation’s Grand Piano Model D thrown in, worth US$79. It’s a very convincing Steinway Model D sample set that uses the UVI engine to deliver great articulation and versatility.

Both come pre-mapped to the Keylab for instant physical control over things like effects, mallet bounce, EQ, and velocity curves.

CONTROL CENTRE

There are a number of ways you can control the parameters and sound of your Keylab 88. Firstly, when used as a standalone controller, Arturia’s MIDI Control Centre application allows you to tailor which MIDI CC numbers are attached to the physical controls, and customise their response to suit. You can store these presets in the software, and sync them to the Keylab’s Working Memory when you need them. The keyboard comes pre-mapped to specific CC values used by Analog Lab. On the Keylab controller there are a further 10 presets for MIDI parameters, where you have much of the same control: velocity and aftertouch curves; fader and knob modes; keyboard splits; MIDI

Channel send assignments; minimum LSB and maximum MSB, and much more.

The final level of control is in Analog Lab itself.

Out of the box, the physical knobs are preset to Filter Cutoff and Resonance; LFO Rate and Amount; a couple of FX parameters; and four other main parameters. You can change any of the parameters to your own selection from a drop down menu on the Analog Lab interface. A nice feature would have been to be able to push down on the encoders and change their parameter assignments on the fly.

The knob controls don’t have end points, so they automatically pick up where the software control is.

The outer markings around the knob ‘fill up’ blue as you turn the control. I found I needed to crane over the laptop to see that onscreen colour feedback while playing the keyboard. The GUI looks lovely, but some parameters aren’t glance-able, and there’s no full screen option to enlarge the visuals.

The faders are typically preset to control filter and amp envelopes, but can be reassigned like the knobs. The physical fader position is represented by a translucent ghost version in the GUI. Once the spectre picks up the position of the GUI, everything falls into sync. Compared to the knob feedback, the fader visuals are spot on.

Lastly, when connected to Analog Lab, the pads are set up to play chords by default. You can change the root and type of the chord easily from the software. These are great for whipping up compositions, or squirting out fast, glitchy chord patterns. And the LCD readout shows the exact MIDI numeric value the pads are currently outputting, giving you great feedback if you’re using pressure to control effects or aftertouch.

Parameter encoder during your set. From there you can assign 10 of those as snapshots, which are linked to the 10 pushbuttons on the Keylab for access at any time.

When you browse sounds, you can filter the options by instrument, type, characteristics, or just browse your favourites or user-defined presets.

And if you own any of the Analog V Collection synths, you can go in and tweak the settings on the synth’s GUI. I found a couple of quirks with the software.

For one, it only ever displays a 61-note keyboard in Analog Lab. Even though the GUI can stretch almost the full-screen in height, it never goes wider. This is only really an issue when you’ve assigned chords to pads at either extremity of the keyboard and want to see what notes are in it, a tiny hiccup. And while Analog Lab was really stable by itself, taxing the computer by using a number of applications at once made mine pinwheel occasionally. Not something I’d be doing in a live situation.

While you can get bog standard keyboard controllers, many these days survive on their ability to integrate into existing systems. The old mode of just providing a few keys and a handful of general MIDI controls aren’t going to cut it. If a controller won’t map — preferably automatically — to my system, I’m ditching it. Arturia’s got that pretty well licked. Its MIDI Control Centre app can take the Keylab 88 from humble piano workhorse to an all-singing, all-dancing circus troupe. Even if you venture outside the Analog Lab paradigm.

MODUS OPERANDI

There are three modes of operation in Analog Lab: Sound, for single instrument presets; Multi, for dual or split instruments; and a Live mode, for sequencing multiple patches. In Multi mode, the keyboard split is easy to set by dragging the upper or lower values of each in the software. You can drag ’n’ drop any sound into the Live mode browser, where you can set up 128 presets in song orders and browse through them with the AT 79

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