BIG & SENSITIVE Audio-Technica’s GroundBreaking XL Condenser
ZOOM H6 Slightly Bigger, Much Better
RICH COSTEY’S DISTORTED PERSPECTIVE Mixing Chvrches’ Debut Album
THINKING OUTSIDE THE BOX Stripping The Console
SHARE TO WIN! A NILE RODGERS MASTERCLASS
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Share To Win! Anyone Can Enter – Just Share The App To Win!
A NILE RODGERS’ MASTERCLASS Nile Rodgers is an absolute chart topper. He drove CHIC to the top many times over, before delivering hits for Diana Ross, David Bowie’s Let’s Dance and of course, Madonna’s absolute smash Like A Virgin. Not just a force of yesteryear, Rodgers once again demonstrated his deft touch, helping Daft Punk’s Random Access Memories storm the charts. We’re giving a few lucky readers the chance to meet the man. Before each show Nile Rodgers is giving a masterclass on songwriting, guitar playing, recording and live performance. We have two double passes to give away for the following Nile Rodgers masterclass and CHIC shows: Perth at The Astro — Sunday December 8th, 2013 Melbourne at Billboard The Venue — Wednesday December 11th, 2013 Brisbane at The Tivoli — Sunday December 15th, 2013
Each double pass is valued at $200. And winners will be drawn and notified on Friday December 6th, 2013. So share the mag to enter, and nominate your city of choice, and you could be picking Nile Rodgers’ brain about his hitmaking picking style. The competition is open to anyone that can get to those venues by those dates. And if you don’t win, you can always purchase tickets from nilerodgersandchic.com, but hurry, they’re selling out fast.
ENTER NOW AT www.goo.gl/1XxVT
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REGULARS
Editor Mark Davie mark@audiotechnology.com.au
ED SPACE
Publisher Philip Spencer philip@alchemedia.com.au Editorial Director Christopher Holder chris@audiotechnology.com.au
Swinging Expectations
Art Director & Design Dominic Carey dominic@alchemedia.com.au
Text: Mark Davie
Additional Designer Daniel Howard daniel@alchemedia.com.au Advertising Philip Spencer philip@alchemedia.com.au
We like artists to be unhinged — Hendrix, Winehouse, Cobain — swinging between emotional extremes and reflecting on life through a kaleidoscopic lens. It’s a creative asset in the business of music-making. And it lends a reality to onstage personas that would otherwise feel fake — their emotional purgatory like a penance for the spotlight.
What you’re left with is trying to maximise the value of the artist’s small budget to deliver something you can be proud of, without falling below the minimum wage line yourself. All the while trying to convince them the best money they’ll spend is on pre-production — a hard sell to anyone who thinks a recording studio is for recording.
But the manic types aren’t always easy to work with. The same drive that makes them unwilling to accept the status quo, a fundamental tenet of their creativity, can manifest in a dictatorial/micro managing work ethic. Sometimes they have the talent to pull it off. But more often than not, when it comes to the technical side of things, a producer and engineer can be the difference between a clear execution of their vision, and a muddy one — the Kramers, Ronsons, and Albinis on the other side of the window.
This isn’t anything new, you could label it ‘the plight of the indie engineer’ but that would be a bit trite.
The record industry usually vets the worthwhile eccentrics through a fairly simple equation. Take record sales, and pit them against the cost of putting up with crazy demands, accounting for the viral infectiousness of Miley Cyrus-like stunts, of course. And to strike on an artist’s promise from the getgo, record companies tend to put a high priority on matching them with an appropriate producer and mix engineer — insurance on the investment. It can be for any number of reasons that the shoe fits. A level head that will bring decisiveness to the process; known for a particular sound that would suit the artist’s repertoire; excels in a particular relevant style; or all of the above and more. Outside of this infrastructure — and with more independence in the business, the place we’re often finding ourselves in — it can get harder to distinguish the outright loonies from the worthwhile eccentrics. We all have different definitions of ‘worthwhile’. Some might say money in the bank is enough, others might be looking for a sense of satisfaction to tolerate headaches— a job well done and songs to match. But often, it’s a bit of both. There’s little return besides a daily rate or an agreed upon fee, and usually not much money to go round. There’s no guarantee of a longstanding relationship, and points are completely worthless when there’s no record sales.
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This economic hamstringing is fine in theory. Demos would be listened to, the budget assessed, and a frank conversation about what would be possible is had. If they want to smash out five songs in a weekend, fine, just don’t expect it to have the sonic complexity of Pet Sounds. A schedule is agreed upon, the job done, and everyone would walk away satisfied with an equal understanding that what was achieved aligned with expectations. But the reality is: producers and engineers are battling with competition from cut-price, EP-in-a-weekend deals. And with a growing shift to DIY there’s a lot more knowledge out there, but not necessarily experience. Which means heightened expectations without the understanding of what’s required to achieve sophisticated results. So instead of resolving to live by experience and call it as you see it, the temptation is to try and leverage whatever experience you have to beat the others at their own game. EP in a weekend? Sure, I can do that. Quicker, smarter, better. But problems mount when expectations aren’t curbed in the face of crippling economic circumstance. You can end up pouring everything into a project, working later, longer, all to try and impress a client that is incapable of being impressed. And instead of them being grateful for the exertion and fitting within their budget, they wonder why the result isn’t the equal of something you spent five times as much time on because there was five times as big a budget (and don’t have the talent of a Hendrix, Winehouse, or Cobain). And if you’re really unlucky, the malcontent might even threaten your livelihood; to denounce you publicly. It happens. So if you can’t identify the unhinged swinging your way, make sure you’re clear on expectations.
Accounts Jaedd Asthana jaedd@alchemedia.com.au Subscriptions Miriam Mulcahy mim@alchemedia.com.au Proofreading Andrew Bencina Regular Contributors Martin Walker Michael Stavrou Harry Irvine Paul Tingen Graeme Hague Guy Harrison Greg Walker James Roche Greg Simmons Tom Flint Robin Gist Blair Joscelyne Mark Woods Andrew Bencina Jason Fernandez Brent Heber Gareth Stuckey
Distribution by: Network Distribution Company. AudioTechnology magazine (ISSN 1440-2432) is published by Alchemedia Publishing Pty Ltd (ABN 34 074 431 628). Contact (Advertising, Subscriptions) T: +61 2 9986 1188 PO Box 6216, Frenchs Forest NSW 2086, Australia. Contact (Editorial) T: +61 3 5331 4949 PO Box 295, Ballarat VIC 3353, Australia. E: info@alchemedia.com.au W: www.audiotechnology.com.au All material in this magazine is copyright © 2013 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. 31/10/2013.
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GENERAL NEWS RIGHT OUTTA THE BOX TBA | www.apiaudio.com
Automated Processes Inc. (API) could be onto a winner here. The company has announced the latest addition to its line of analogue consoles called The Box, describing it as a ‘project console’. The Box is specifically designed for audio professionals with project or home studios who require a smaller format console, but still want that elusive ‘big console’ sound – in this case it’s safe to assume that means a big API console sound. With this result in mind The Box features the same circuitry and performance as the company’s existing Vision, Legacy Plus and 1608 consoles. The Box promises an easy, turnkey solution for recording and mixing, for people who record a few channels at a time, but want the warmth and punch that larger consoles can deliver. Fundamental to achieving that is the API stereo compressor on the program bus. That’s in addition to just four inputs, full centre section control and 16 channels of summing. It seems a little boutique for a home studio, but you can’t deny the appeal and API may have The Box design formula just right. Studio Connections: (03) 9874 7222 or www.studioconnections.com.au
NEW M-AUDIO MONITORS $719.99 apiece | www.m-audio.com
M-Audio’s M3-8 Studio Reference Monitor features an efficient inline/coax design that provides three-way sound in the same space as a standard two-way monitor (the MF and HF drivers are mounted inline). By delivering sound along the same focal plane, the M3-8 promises improved time-alignment for fatiguefree listening. The eight-inch low-frequency driver and five-inch mid-range driver are made from lightweight woven Kevlar. The one-inch silk dome tweeter offers integrated waveguides to provide increased clarity. In addition to the real wood baffle, the cabinet uses tuned bass porting and optimised internal bracing. Three individual amps provide power to each of the three speaker elements for a total of 220W of Class A/B amplification. A three-band EQ with a switchable low-cut filter allows you to shape the sound to match your listening environment and pinhole-mounted blue LEDs are a neat and easy-to-use visual aide for achieving perfect speaker placement. Full specs and details are on M-Audio’s website where you’ll also see the M3-6 model mentioned, but that’s slated for next year. Meanwhile, we should see the M3-8 available in time for Christmas. Pro Audio Group: (02) 9521 4844 or www.proaudiogroup.com.au
Neil Young is still determined to save the world – the musical part of it at least – but now we’ll have to wait until next year, 2014. That’s the most recently-announced launch date for Young’s own high-quality music streaming service called Pono (which is Hawaiian for ‘righteous’). The details are still to be revealed as to how Pono will deliver high-resolution music over the web except that Young explains it will be sourced from artist-approved studio masters. Like everyone, we can’t wait to find out and hear the results. The best way to keep track of Pono is through its Facebook page. facebook.com/neilyoungpono
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When Sennheiser first released the HD25 headphones, the company probably didn’t consider its design would stick around long enough to coincidentally get a spit and polish re-release to mark the 25th anniversary of Sennheiser making cans – but that’s what’s happened. Talking of cans – the recycled beer kind – the anniversary HD25 edition is called the HD25 Aluminium with the earcups sporting a distinctive, aluminium finish having been carved out of solid lumps of the stuff. Apart from that, the HD25 are a solid argument for “if it ain’t broke, don’t fix it”. Sennheiser Australia: (02) 99106700 or www.sennheiser.com.au
Speaking of headphones, pull on your favourite ripped jeans, best bling and read on. Monster, which has a tradition in high quality hi-fi cable but is now best known for headphones, has teamed up with lifestyle company Diesel, to launch a new range of headphones: the Monster Diesel Noise Division VEKTR collection (yes, these are a fashion accessory and therefore qualify as a collection). The VEKTR headphones – available in Military Green, Black and White, represent the leading edge in today’s merging worlds of high fashion and technology. I think the jury is in: headphones are now a fashion accessory. www.monsterproducts.com
MOTU’s new 8pre interface is a dual-purpose device. The 8pre audio interface provides everything you need to turn your computer into a 16-channel 24-bit/96k digital audio workstation via eight microphone/instrument preamps with 96k analogue recording and playback, combined with eight channels of ADAT optical digital I/O. It can also work in a standalone mode where the digital I/O becomes an A/D converter for the microphone preamps to connect into an existing system without the need for the USB port to be connected or software drivers controlling the 8Pre. Good idea. Network Audio Solutions: 1300 306670 or www.networkaudio.com.au
Going digital ain’t so bad... PRISM’S TITANIC LAUNCH TBC | www.prismsound.com
Prism unleashes Titan, and if it looks kind of familiar, that’s because it is essentially a USB version of its Mt Olympus stablemate, the Orpheus Firewire interface – plus a few tweaks such as the new MDIO interface expansion slot. Otherwise, it offers eight analogue inputs, eight analogue outputs plus S/ PDIF and TOSLink optical digital I/O ports. The optical ports can also be used for ADAT, all up giving Titan a maximum capability of 18 concurrent input and output channels plus dual stereo headphones. Titan has four preamps for microphones and two for instruments, which are automatically selected when sources are plugged in. An assignable rotary control adjusts monitoring or output level controls. The dual stereo headphone outputs have separate volume controls. A built-in DSP engine provides a fully-featured mixer for every Titan output channel, including ADAT channels. All of Titan’s functions, aside from the front panel monitor and headphone level pots, are controlled from the Titan Control Panel software. CDA Professional Audio: (02) 9330 1750 or www.cda-proaudio.com
AUDIOTECHNOLOGY: THE MONTHLY APP
MATRIX 2: THE SSL SEQUEL
Lower latency, cheaper and more in tune ...but with all the soul of the original
$29,999 | www.solidstatelogic.com
The original SSL Matrix console appeared in 2008 offering a hybrid production platform – a combination of SSL analogue summing, integration of analogue outboard mic pres and processing (via its software controlled analogue patch system) and advanced DAW control surface. It’s the integrated software-controlled patching of analogue channel inserts that has been most significantly upgraded in the Matrix2 software update. Hardware device inserts can now be loaded directly from the console hardware controls, previously this was done only via the remote browser software, with an intuitive new interface that facilitates loading individual processors, A/B comparison of different processors and building processor chains. The Matrix remote browser software has also been re-designed to provide a new ‘drag and drop’-style interface for loading processors and building chains. Other new features include Partial TR setup save and import (which allows selected parts of the console setup to be saved and imported as setup templates), new Preset insert matrix scenes, Preset insert naming tools, automatic dB readout for Pro Tools users and new DAW templates for Presonus Studio One and Ableton Live. Matrix2 will ship in December 2013 while it’s a free software upgrade for existing users.
The new AudioTechnology App has arrived on the iPad and will soon be ready for Android tablets. All the latest news, reviews, features, columns and tutorials every month for next to nothing. Stay tuned for more news at audiotechnology.com.au or like us on Facebook.
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Amber Technology: 1800 251367 or www.ambertech.com.au AT 11
SOFTWARE NEWS SATURATION POINT
$149 (Native) | www.waves.com Waves Audio and Abbey Road Studios have teamed up again, this time to create the J37 tape saturation plug-in, a precision model of the very tape recorders used at Abbey Road. The J37 was famously, innovatively used by George Martin and the Beatles for the <Sergeant Peppers> album. With a variety of user-adjustable controls including Tape Speed, Bias, Noise, Saturation, Wow and Flutter, the Waves: Abbey Road J37 recreates the sonic signature of the original machine including, believe it or not, modelling three exclusive oxide tape formulas by snaffling the last of the tape stock from the EMI archives (specially developed by EMI during the ’60s and ‘70s, each formula has its own unique frequency response and harmonic distortion behaviour). As well as the controls mentioned above the J37 tape saturation plug-in features Comprehensive Tape Delay, Sync, LP and HP filter controls. The J37 is Native and SoundGrid-compatible and isn’t (so far) included in any Waves bundle. Both formats are now available at special introductory pricing of $149 (Native) and $249 (SoundGrid). Sound & Music: (03) 9555 8081 or www.sound-music.com
ANOTHER FRENCH VINTAGE $10.49 | www.arturia.com
Arturia has announced the iSEM, its second subtractive synthesizer recreation for iOS following on from the iMini Moog app. Like Arturia’s Oberheim SEM V desktop software solution, iSEM uses proprietary TAE (True Analogue Emulation) technology to reproduce the analogue warmth and ingenious interface of the vintage Oberheim Synthesizer Expander Module (SEM), but brings it all “kicking and screaming” into this day and age by making the most of the latest developments available to its iPad host. It’s Audiobus compatible, however Arturia has been quick to take advantage of iOS 7’s built-in Inter-App Audio, allowing iSEM (and iMini) users to send MIDI commands and stream audio between apps on the same device – potentially an Audiobus killer, but no doubt the folks at Audiobus are hard at work on a riposte. The iSEM main screen mirrors the original synth with all the extra iOS tricks and tweaks hidden under the vintage GUI. We’ve put our hand up for a review unit and after the Arturia iMini the crew in the office will be squabbling over who gets it. Price at the Australia App Store is $10.49 and it’s a 60MB download. CMI Music & Audio: (03) 9315 2244 or www.cmi.com.au
Novation is starting to get a little confusing with all its ‘Launch’-themed controllers and apps, but the range is growing, so hang in there. Now Novation has announced two new hardware controllers aimed at live performance, the Launchkey Mini, a compact new mini-key instrument/controller, and the Launch Control. Launchkey Mini works with the new Novation Launchpad and Launchkey iPad apps, and most Mac and Windows-based DAW software. Likewise, Launch Control is a robust, compact controller with 16 assignable knobs, eight three-colour launch pads, and four function keys. Innovative Music (03) 9540 0658 or www.innovativemusic.com.au
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Waves has announced the ‘long-awaited’ release of AAX Native plug-in support, making the whole thing sound as if it’s been some kind of exciting, eagerly anticipated event rather than the last chapter in a long story of many Waves users rapping on the company’s windows and demanding to know when Waves will pull its finger out and get on the AAX Native bandwagon! Waves AAX-compatible plug-ins are available at no additional charge for all products covered by the Waves Update Plan. Otherwise you can opt back into the plan any time and get the same result. Sound & Music (03) 9555 8081 or www.sound-music.com
If API’s (Automated Processes Inc) new The Box console, mentioned elsewhere, doesn’t tempt you, Universal Audio has released the API Vision Channel Strip Plug-In for the UAD Powered Plug-Ins platform and Apollo audio interfaces. Featuring five API modules the API Vision Channel Strip plug-in offers owners of Apollo audio interfaces and UAD-2 DSP Accelerator hardware a faithful channel strip emulation and the ability to impart their tracks with the same focus, punch, and harmonic detail as the original console. You can buy the API Vision Channel Strip Plug-In from UA’s Online Store for US$299, and it’s part of the new UAD Software v7.3 release. CMI Music & Audio: (03) 9315 2244 or www.cmi.com.au
From a Gibson press release, buried within a jumble of legal gobbledegook, is the news that Gibson is making a serious play to purchase Cakewalk, the developer of Sonar. The language is quite amiable and it seems to be just a matter of thrashing out the finer details before Roland Corporation hands over Cakewalk, which will then come under the mantle of a newly-created company called Tascam Professional Software. Cakewalk will continue to operate out of its Boston headquarters as an independent division and apparently won’t lose any of its branding. Distribution remains the same as well. Intelliware Australia: (02) 9981 8088 or www.intelliware.com.au
CONTROL YOUR DOG www.machinewerks.co.uk
The Black Dog is a three-piece, UK-based band that with various line-ups focused around founding member Ken Downie has been around since the early 1990s and is close to legendary in the techno and dance music scene – even credited with inventing ‘intelligent dance music’ or IDM (now, now, before you say anything…). The lads are enterprising, starting their own record label and such, but the latest Black Dog project is more ambitious. After launching a company called Machinewerks, they’ve embarked on designing, building and eventually distributing the ultimate USB/MIDI control surface for DJs and electronic musicians, since apparently nothing currently on the market was up to snuff. The result is the CS X51 controller and the next step is full production. For financing this phase, money is being sourced from crowd-funding through Kickstarter and the target of £27,000 has just been reached. However, there’s still room on the Black Dog bandwagon and if you’re interested in a piece of the CS X51 action, or curious about the controller itself, the Machinewerks Kickstarter page is full of information.
WELCOME TO MASCHINE
pure energy
TBA | www.native-instruments.com
Maschine Studio is Native Instruments’ flagship version of its hardware music production system. Physically bigger and (as you’d expect) better than its predecessors, NI promises that Maschine Studio redefines hands-on control. Two large, high-resolution colour screens show waveforms, patterns, scenes, NI product visuals – meaning the graphics of included NI software such as the Solid Bus Comp, plus any mixer and meter levels in full detail. A grid of 16 multi-colour pads allow for real-time input of beats. Eight display knobs, edit buttons, and a jog wheel with LED indicator will give tactile control of Maschine’s sequencer, instruments and effects. A dedicated level section delivers clear metering and monitoring. A built-in stand lets you adjust the performance angle. Maschine 2.0 software delivers a tag-based visual browser, unlimited groups and insert effects, side-chaining, and a mixer – all directly accessible from the hardware. Maschine 2.0 hosts any VST or AU plug-in and integrates into any studio setup as a plug-in in all major DAWs. The software also has full multi-core processor support. Maschine Studio comes with an 8GB library. CMI Music & Audio: (03) 9315 2244 or www.cmi.com.au
The New Neumann KH 310A Three Way Studio Monitor is in a class all of its own. With state of the art technology, the KH310A delivers cutting edge performance and extreme accuracy at surprisingly high reproduction levels . The result is a sweet spot that only Neumann can deliver.
For more info on the Award Winning Neumann Range, or details of your nearest stockist, call 1800 648 628, email sales@sennheiser.com.au or visit neumann.com
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LIVE NEWS BROADCAST SD5 www.digico.org
Digico has added a broadcast application specific extension which allows the SD5 to become the SD5B. As with the SD5, the SD5B’s work surface is a low noise, heat dissipation work surface benefitting from Hidden-til-lit (HTL) technology, however its five digitally driven full colour TFT LCD screens, three of which are touch sensitive, have a new configuration that allows easy access to single or multiple users. There are also two interactive dynamic metering displays (IDM) and quick access buttons are positioned down the left side of the channel screens for fast and easy navigation. Incorporating the master screen into the work surface design caters for complete user feedback,
but maintains a lower profile meter bridge for clear visibility of those on stage for the user. As standard, the SD5 comes with a 2Gb fibreoptic system, plus 56 console-to-console tielines. There are three redundant MADI ports and local I/O includes eight microphone inputs, eight line outputs and eight AES I/O (mono). The SD5 has 124 input channels with 56 configurable busses. For broadcast applications, the SD5B also includes dual solo busses for PFL and on-air soloing in mono, stereo LCRS and 5.1 and an LR/ LCR/LCRS/5.1 master bus. Group Technologies Australasia: (03) 9354 9133 or www.grouptechnologies.com.au
LINE 6 DIGS EARTHWORKS TBA | www.line6.com
Line 6 probably isn’t the first brand you’d think of when it comes to radio mics, but the company has been steadily developing its range of wireless products and the latest offering, the V75-40V handheld, comes with an Earthworks WL40V hypercardioid capsule. Designed to deliver ‘studio-quality’ performance in a wireless microphone, the hand-tuned and tested WL40V capsule promises lightning-fast impulse response, high SPL handling and a wide frequency range. The hyper-cardioid polar pattern provides ‘exceptional’ detail and nuance, resulting in vocals that require little or no EQ. The rest of the V75-40V package includes up to 14
When we ran a review of the Allen & Heath ICE-16 multi-recorder in Issue 93, the unbalanced TS inputs and RCA outputs were worth mentioning, although the verdict was it didn’t cause any real headaches. Regardless, A&H says it has been listening to its customers and has now introduced the ICE-16D, an alternative version of the ICE-16 with the ‘D’ denoting that its connections are fully balanced via a brace of D-Sub connectors. You already know all the other features because you’re subscribed to the magazine and you read Issue 93, right? Technical Audio Group: (02) 9519 0900 or www.tag.com.au
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Solid State Logic’s new v4 software for its C10 HD Compact Broadcast Console introduces a range of features and options that significantly expand the capability of the C10, a mixer that promises ‘large console power delivered in a compact, affordable and intuitive package’. In particular, a range of automated features and simplified controls make the C10 adaptable to environments where users of ‘varying skill levels’ will operate the console (no, not you – sit down at the back). Also, the maximum channel count available for C10 has been increased to 160 channels. v4 Software for C10 HD is available now from SSL’s website. Amber Technology: 1800 251367 or www.ambertech.com.au
channels to choose, 24-bit/10Hz–20kHz frequency response, backlit LCD display, a lock-out switch and transmitter naming for identifying transmitters in a multisystem application. The mic has a metal body and comes in a custom hard-shell touring case. The one thing you don’t get in comparison to Line 6’s other handheld wireless products is the built-in microphone modelling – not surprising. Earthworks would no doubt get a bit miffed at any suggestion users would even want mic modelling over the genuine capsule. Australis Music Group: (02) 9698 4444 or www.australismusic.com.au
TC Electronic’s System 6000 signal processing flagship has been around for over a decade in professional recording, mastering and film studios. The recent introduction of System 6000 MkII saw it being used for live transmission as well as post production. Now there’s the Broadcast 6000, a custom-tailored version of the System 6000 MkII optimised for production with broadcast in mind. The new unit comes fully equipped with a wealth of pristine algorithms including TC Electronic’s LM6 Loudness Radar Meter, AM6 Radar Meter and multichannel dynamics processing algorithms. Amber Technology: 1800 251367 or www.ambertech.com.au
Waves has opened up SoundGrid with a couple of handy MADI interfaces. The palm-sized MGB coaxial and MGO optical MADI boxes let you interface any MADI equipment. Allowing you to record, process, and play back 128 audio channels at 48k using Waves and third party plug-ins all to the low latency tune of 0.8ms. You can even simultaneously split that signal to two computers for backup. Group Technologies: (03) 9354 9133 or sales@grouptechnologies.com.au
X32 GOES TO ALEXANDER’S ELEPHANT Gordon Alexander from White Elephant Sound was the winner of the Behringer X32 subscription prize giveaway. It was one of the biggest prizes we ran this year, and Gordon’s had a little bit of time to get settled with his new console. Here’s what he had to say about it: “Wow, this X32 is a great piece of kit! In use it is as straightforward and functional as any digital desk these days could hope to be: Well laid out with all the most used parameters at your fingertips, lots of routing options and very usable mic preamps and effects. I’m using it in a studio environment and would have liked higher resolutions, but 24-bit/48k will do just fine for most things. The control surface function integrates well with both ProTools and Nuendo and I’m loving having faders to control my DAW rather than using my old mouse... This thing is an all round pleasure to use. Thank you very much AudioTechnology and Galactic Music for offering such a great prize!” We’re chuffed that the prize has gone to such a deserving reader, and we hope it will get use for years to come. Thanks again Galactic Music.
THE GROUNDBREAKING
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FEATURE
How Rich Costey and Chvrches learnt to stop worrying about the meters and trust their ears. Story: Mark Davie Chvrches Photos: Dexter Cornelius
The Sound City movie really pissed Iain Cook and Martin Doherty off. “They just blamed ProTools for everything!” said a clearly outraged Martin. “For the death of creativity and spontaneity,” chimed in Iain. The pair are two thirds of Glaswegian synth pop band Chvrches who just dropped their debut album The Bones Of What You Believe. The tour cough had caught up with singer Lauren Mayberry, but a producer herself, it’s likely she would have been weighing in too. “It was like they don’t get it, but in an equal and opposite way,” commented Martin. “Sure, it’s not just about digital, but anyone who says it’s not useful in some way, and that you can’t strike a balance marrying beautiful analogue signal paths to the almost endless possibilities of digital is shortsighted and a fool.” The pair aren’t proselytising on behalf of Avid. They don’t even use ProTools. Nuendo and Cubase in the studio, and Ableton Live on the road, are their tools of choice. But that’s exactly their point. There are so many different music-making tools in the digital environment, so resorting to name calling is to miss the creative forest for the trees. “You ask any huge rap producer and he’ll probably tell you he uses Fruity Loops,” continues Martin. “Really man? Fruity Loops? C’mon! But that’s the thing, it’s your process, your personality, the way you work. The software you use is important, but more importantly, it’s how you use it. And the guys using FL Studio are doing incredible things that are really difficult to do in a classic, linear multi-tracking situation.” “We always talk about how gear and software inform the writing process,” said Iain. “Because a lot of our ideas come from a drum sound or sample or a synth sound, and that technology inspires our songwriting. Maybe we should try FL Studio.” AT 16
MANY DAWS TO ENTRY
The pair have been serial DAW hoppers. Before Nuendo and Cubase, Martin used Logic, back when Emagic still owned it. He eventually switched because he didn’t want to pony-up for a new Mac. These days it’s Apple laptops all the way, but they haven’t managed to gel with Logic again. Though when Chvrches crossed paths with Sigur Ros at Rich Costey’s studio in California, he was mixing their record in Logic because that’s what they recorded it in. Which got the wheels turning again. “If those guys can do their whole album in Logic,” said Iain. “There must be something weird about it.” It’s actually a working philosophy of Costey’s: to mix a record in the same DAW it was created in. “I’m a big believer that whatever program somebody uses to record their music, is the one it should stay in,” he said. “Most people who mix records use ProTools and I do as well. “I got rid of the TDM system and switched to the native card, which enables you to boot up whatever program you want and still access loads of analogue outputs. Chvrches recorded that album in Cubase, so I mixed it from Cubase.” (opposite) Lauren Mayberry provides the sweet, pure melodies that holds Chvrches’ distorted dish together
One reason Costey likes to keep it native is because more and more he’s finding himself mixing albums while bands are on the road or on the other side of the world. So it pays to have the same session as the band in case they’ve forgotten to finish a line, or didn’t backup a part properly in the haste to get to the airport. BREAKING BAND
He gets a bit of masking tape out and just starts taping up the meters and draws a reading where it’s nicely out of the red
”
Case in point, Chvrches. The band had only just finished tracking in Iain’s basement studio in Glasgow and already their management team was pushing them out on the road. “When a band breaks on the internet, and not in an individual territory or one territory at a time,” said Martin. “You’re suddenly in a really fortunate position where people want you to be everywhere at once.” There were even “a few things we had to do on the hoof, because we had to finish the album on tour,” explained Iain.
Iain and Martin are not only savvy musicians — having had successful indie band careers prior to Chvrches — but engineers too. “In both our own band contexts, we were always the guys who were put in charge of audio technology,” said Martin. “We both learned how to use computer audio technology from an early age. I think the first computer I had was an Intel Celeron 800. I couldn’t afford the P4. I was broke.” Watch: The Mother We Share from Chvrches’ The Bones Of What You Believe
“P4?” said Iain. “Try the P1 mate. That was my first one.” Iain had recorded a couple of previous band’s albums in his studio, which is set up primarily for his TV and film post-production work. And the band mixed two of the tracks on the album, Tether and By The Throat. But you can’t fight fame if it’s knocking at the door. So as the band stepped out on an exhaustive tour schedule, they had to find a mix engineer. And not just any mix engineer. “Handing it off was a very difficult thing to do, but we just ran out of time,” said Martin. “Not to be conceited, but we felt we had enough knowledge in the locker to see it through to the end. And if we were going to get someone, we wanted someone world class. Specifically someone who would take it and look at it more as a rock recording than a dance or electronica recording. So then you’re really talking about
one of three or four guys in the world — Rich [Costey], Flood or Alan Moulder. We were lucky Rich was free.” Rich mixed Recover as a test track, and there was no question he was the right mixer for the job. In all, Chvrches were only able to attend a single day-long mix session. With “their tour bus parked out front,” said Costey, they recalled as many songs as they could get through. They spent the rest of the time watching vicariously through a live stream of the console, and communicating via Skype. IN THE ORIGINAL BOX
“The way the Chvrches album was mixed was not unusual,” said Costey. “I’m often doing mixes for an artist who is on the road somewhere. Occasionally I’ll be mixing and there’ll be a keyboard or vocal part missing from the session. By using the same session they can easily send the part right from their laptop, and I open up the session and it’s fine. They don’t have to bounce anything out. “We also keep all the plug-ins running live. Whatever ones they’ve got, I’ve got. So if we’re doing filter sweeps, or echo feedbacks, we can interact. They can do a pass of delay feedback and send it to me from wherever they are. “The other reason I use the same DAW is I think each program has a peculiar sound to it and if an artist has built up their character and identity around using this particular program, not using it goes down the road of diluting what they’ve built up. “For example, Logic has a very particular sound to it. You can pick it out a mile away. I worked on the first Foster The People album and for Mark [Foster], Logic is like an extension of his body, he’s so comfortable and quick with it. That album has a particular sound because almost all of it was done in Logic. “I started this approach when I was mixing a record by The Shins for Greg Kursten, who produces a lot of pop records. I found bouncing out Greg’s tracks to ProTools wasn’t really working because he had so many different aux channels running into one another. With his drum sounds, when you bounce out the kick or the snare, you’re not hearing how all the different compressors react with the whole kit. So to me it didn’t sound right unless I just ran his sessions completely.” BUILDING THE CHVRCH
Chvrches is built around the perfect pop marriage of Lauren’s pure, almost nymph-like vocal delivery sitting easily atop an aggressive, but spacious rhythm-driven synth landscape. It didn’t start out that way though. Martin Doherty: It happened organically. Initially, I was singing on most of the demos, but they were just the sketches coming out of our heads. Iain played me some of Lauren’s stuff and we talked about a totally different timbre for the AT 17
IC: We’re always figuring out when not to fill up the space. Take Depeche Mode. A lot of the time they only push limited frequencies in their arrangements, but there’s so much room for the song. MD: Or When Doves Cry by Prince, that’s more or less drums and vocals. IC: No bass line.
Iain Cook plays on his Moog Voyager, with Ableton Live in the background running everything from playback to the entire band’s monitor mix
recordings. She came in, tried some stuff out, and we had a proper moment in the studio where we knew it was cool. When we realised how well it could possibly work we started writing songs together with Lauren and she became an intrinsic part of the creative process. She allows us to push things harder because we have something to bring balance to an audio recording — a voice that’s sweet and a pure tone. As long as her voice holds that top line, the song can still be pleasing to the ear. That was something that clicked almost instantly when we heard her sing.
AT: How have you done that? IC: By layering up wide sounds with three oscillators at a really wide pitch base. Distorting things, using a lot of arpeggiators, making four or five passes with different voicings of chords and then chopping them up and stacking them into a sampler so you’ve got this huge sound that’s triggered with individual hits. We’re giving away all our secrets!
There are some ‘brick wall’ elements on our album but the classier you get the more you start to understand and realise that it’s about where the space is rather than filling up the space.
MD: When you don’t have a drummer accenting all the divisions between the kick and the snare on a hi-hat, and hitting the toms all the time, all this high end suddenly appears which you can either fill up with shite or let breathe. Or you can find new ways to accent those rhythms — a piece of percussion, some lines of deliberately chopped vocal.
Iain Cook: It’s a really hard thing to learn. Sometimes you go down the line of filling up an arrangement until there’s no room left, and you’ve lost the original energy it had. In an altrock context you fill everything up with cymbals and loud guitars. For this project, it was about finding ways for synths to sound big like that.
We’re all about crude sampling, and place those samples in the triplet or eighth-note divisions. We put a lot of work into how the rhythm tracks are composed and it usually starts with a big dirty kick drum and a driven snare, which is usually three or four sounds morphed or amalgamated together.
MD: Realising that was a mind-blowing moment. And so many electronic recordings are clean and soul-less. It’s part of the reason we don’t use a step-sequenced or scene-based approach, because we want to retain some of the roughness of an alternative rock group recording, which is our background. The idea that everything is played, and some things are a wee bit behind. You can still use dance production and styles — like the whole song just being different notches on a filter or one keyboard line — but there has to be that organic and played human element for it to excite us. The same thing goes for tonality. There was a really inspiring interview with Adrian Utley from Portishead we watched when we started on this project. He said he had no interest in sounds that are sonically perfect or beautifully in tune, and that appealed to us. IC: Soundtoys’ Decapitator and Devil-Loc had a big sonic stamp on the record. DIRTY SECRETS
In fact, distortion was liberally stamped on all aspects of the record, which is somewhat of a specialty for Rich. Rich Costey: “The main directive was to try and dirty it up and keep it from getting too clean. They wanted it to be powerful in a way that most people don’t think of synthesiser albums sounding. Most clients I get want their albums to work on many levels, but shiny-ness is usually not one of them. AT: What are the fundamental reasons behind dirtying it up? RC: In the case of Chvrches, they wanted the music to have a slightly more punkish approach and more power than a traditional synth record. When you add distortion to the synths and
SYNTHS ON THE ALBUM Martin Doherty: I’ve grown up with software synthesisers, trying to push computer-based DAWs as far as possible and learned how to use hardware afterwards. But as much as the software recreations are amazing, and invaluable for people that are working on a budget, they never come quite close to the sound of the real thing. Iain Cook: There are a few software synthesisers on the album, but that’s partially because we couldn’t afford better hardware. Like the Roland Jupiter 8. Back home we could pay £4000 for one. MD: And the Yamaha-CS80 and the ARP2600. All those Arturia re-creations are really useful. AT 18
IC: Even so, comparing the Arturia Minimoog to a real one, it’s close but doesn’t have the same punch or feel. Though it’s really difficult to put your finger on what makes the real thing sound better. The three hardware synths we use live are a Moog Voyager, a Roland Juno 106, and a Prophet 8. We had the chance to meet Dave Smith in San Francisco a couple of months ago. He showed us around the offices and gave us a tour of the Prophet 12 and came to the show and drank all our tequila!
MD: We bought the Prophet 12 the next day. How can you get a demo off that guy and then not be blown away to the point where you just need to
own one? But it’s stuck in America right now, we can’t get it until we go back. IC: We recently bought a Dave Smith Instruments Mopho X4 and a wee Moog Minitaur bass synth. We’ve got a Teenage Engineering OP-1 which is a lot of fun. MD: And a Dave Smith Voyager One Tempest, which is just about the most versatile drum machine I’ve ever used. It has so much character. IC: It mixes the early sampling world like the Linndrum and classic 808-type sounds, with a modern operating system. And we’ve got a Yamaha DX7 in the studio as well.
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AT: Have you been through a lot of saturation devices over the years? RC: In any given case the type of distortion has a huge effect on the presentation of whatever it is you’re distorting. But types of distortion to me are more like fads. I get excited about using one on everything for a while, and then I move on to something else and use that on everything. I had an EMS Synthi A for a long time and went through a period where I ran almost everything through that. It’s the world’s most expensive distortion box. The Synthi A doesn’t come with a keyboard. It’s a nightmare even trying to get that thing to play an octave in tune, so it’s basically just for external processing. I still occasionally use a Symbolic Sound Kyma system for distortion. AT: How did you come to use the Kyma — it’s not something you often come across in pop record mixing? RC: I’ve had the Kyma maybe eight or nine years now. I remember reading about it and it seemed like a really difficult piece to get your head around, but if you spent time with it you could probably get it to do almost anything. That sounded really appealing to me, so I went and got one. It’s similar to Reaktor in that it has modules and sub-modules that you can route into one another to create samplers, keyboards, processors, whatever you want. It’s incredibly deep. It has its own sequencer but the sequencer isn’t for recording notes, it’s for recording changes in the processor over time. I have to admit I have been a bit too busy to ever really master it, but I can fumble my way around the thing enough to make pretty reasonable use of it in a session. VOCAL PRECONCEPTIONS Rich Costey’s studio is a treasure trove of gear that helps deliver the intent of his artists. At the centre of the control room is an SSL 4000 series console, which he’s just switched out for a K. The vintage Universal Audio 610 console is one of his prized pieces. But you wouldn’t sniff at the Fairchild 660 duo either.
drums, and put everything back together, they start to live in a state they wouldn’t otherwise have. Things are slightly less discrete sounding, on purpose. I’m sure it’s partly due to the band being constantly on the road too. They probably wanted their record to sound a little more familiar to them, like what they’re hearing coming off the stage every night — to have some of that energy. AT: What are some of your dirty tools? RC: A go-to plug-in on that record was Decimort from D16 Group. It’s like a bit-crusher, but it’s got some filters and a built-in distortion. It’s supposed to mimic the sound of an E-mu SP-1200 or something. I don’t know if it sounds anything like an SP-1200, but it sounds good. I would use that and Soundtoys’ Decapitator a lot. I’ve also got an ARP2600 I would occasionally run stuff through because the preamp sounds really nasty. Occasionally we’d clip a couple of Neve channels, but that’s mostly it. I used a lot of plug-in distortions, because we upload a mix and then we might not hear from them for a couple of days, so we wanted to keep things easy to recall.
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Having never worked with Rich before, Chvrches had some big-name preconceptions of the talented mixer that probably wouldn’t be much different from those anyone else would hold — vintage gear piled to the ceilings, and perfectly honed processes. Their in-studio day was a bit of an eye opener. IC: We had this idea that Rich used the most expensive outboard vintage gear. He does have a lot of that stuff, but much of the time we’d ask him how he got a particular sound, and it would be some freeware distortion plug-in he just downloaded. What drives that guy is just what sounds good. His whole approach to this album was quite surprising to me. It’s amazing the way he throws anything at it and doesn’t give a f**k what the meters are saying. MD: In the studio with him and looking at some of the meters, we were like, ‘Is that cool?’ IC: Have you actually taken 16dB off that whole mix there? MD: Then he gets a bit of masking tape out and starts taping up the meters and draws a reading where it’s nicely out of the red. Another thing we couldn’t fathom before we got there were his vocal sounds — it sounded like a 100 grand. Then we got there and saw his vocal chain… and it’s worth 100 grand! RC: I don’t know if it’s quite that much, but you’ve got to get the vocal right. If you can’t get the vocal right, people are going to stop hiring you pretty quickly. AT: So what did that vocal chain look like for Lauren?
RC: The vocal will first go through a Lang EQ, and then the Avalon 2055, then split into three channels on my console. When you set the frequency to the uppermost limit on the Lang, which I think is 20kHz, and turn up the top end, it seems to add harmonics on top of the vocal without affecting the content of the vocal. In other words, you can sort of brunt it up without it getting more percussive on the Ss or the Ps. Then each split has a different piece of outboard gear. One channel has an UREI 1176, the other channel will sometimes have a Fairchild on it, sometimes just an API EQ, sometimes a Teletronix LA2A, and sometimes a Universal Audio 610 module from our vintage UA 610 desk. It depends on the song. The third channel is often a Standard Audio Level-Or 500 series module. I wouldn’t always use it, but it’s there if I need it. Then I mix them together. I run a modified Dolby channel on a parallel which also brings out the top end of the vocal a little bit without it getting too ess-y. I also have another compressor, a Neve 33609, that all the vocals are paralleled into. It glues them together a bit.
of a mess. You can run a dirty synth and a dirty vocal, it just depends on the style of music. These guys are writing pop songs with an old-school punk energy, so I tried to get the tracks to sound really clobbering and have her vocal in there, but not too loud. They were always mindful of making sure she never got too up in front of the track. Notably, nearly all the vocal delays are plug-ins, except for a little bit of TC Electronic D-two. I’m really into Fabfilter’s delay plug-in, it’s awesome. And there are a few passages on the album that have really effective filter delays using Soundtoys plug-ins. We did use all hardware reverbs —
“
The mix bus is terrifying! It’s something you don’t enter into lightly… It starts out with a couple of Fairchild 660s…
”
AT: Are you compressing them much, or is it more about the combination of sounds?
It starts out with a couple of Fairchild 660s, one on each side. Then it goes into a Shadow Hills mastering compressor. Sometimes with Chvrches there would be a pair of Esoteric Audio Research 660s, and then into a Maselec EQ into Millennia EQ, which would sometimes be swapped out for the GML EQ. And then for most of the Chrvches record it would go into a pair of 610 console channels to boost the bottom end. The Fairchilds are not compressing at all, it’s just about the sound. It’s hard to explain. The Fairchild does the same kind of thing the 610s do. I’m not going to them for a particular compressor or EQ, but it does this weird thing where it makes the music sound more important. It’s the only way I can describe it. And then we’re compressing a bit on the Shadow Hills depending on the song. Some of the songs I would hit at a harder ratio and some would be a ratio of 2:1. AT: Sounding ‘more important’ is a little vague. Is there any factor you could put your finger on?
RC: It’s more about the flavour of the devices because, honestly, most things come in reasonably compressed these days, no matter who the artist is. They don’t usually need a whole lot of level control.
AMS, and Lexicon 480 and 224 reverbs. There are a couple of plug-in reverbs I think sound okay, but hardware reverbs are much better whenever you can get them. Even if they’re cheap and shitty, they have more character than a plug-in.
AT: Was keeping Lauren’s vocal pure, compared to the dirty synths, a focus of yours?
MONEY SHOT
Rich: If her vocal was distorted on top of the background, you’d start to lose the character of what she’s doing and it would start to sound a bit EMA_AT94_HR.pdf 1 10/04/13 10:46
a super high-end power conditioner because you just wouldn’t try this kind of setup without it. If you get even the slightest mismatches between gear you’re going to have loading problems. This whole system has taken a long time to get to this point and everything has been A/B’d to the billionth degree to make sure it’s working.
While the vocal chain might not have topped the 100 grand mark, Costey’s mix bus surely does. RC: The mix bus is terrifying! It’s something you AMdon’t enter into lightly [laughs]. All the gear is on
RC: With digital recording, there’s always a bit of a battle. In many ways it sounds empirically better — if you’re running really good converters and a good clock — than your average analogue recording. But when you have 80 or 90 tracks of digital source material running, then everything starts to sound kinda small and ill-defined to me. The intent of the artist is better preserved in analogue recordings than in digital ones. In digital recordings you’re always trying to enhance the intent. If you listen to Back in Black and it sounds tough as f**k, then you are
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compressing your low end that’s when you begin to lose it. It’s about balance and feel. The drums most certainly get some sort of parallel compression to make them sound bigger but if you start running a lot of drum compression it will change the length of your drums. With this kind of music, everything is designed to pulse in just the right kind of way, and if you start making the kick drum longer through compression, you’re going to mess all that up. The goal of a compressor is to make it sound like someone is playing harder — I think that’s a noble goal. But quite often I feel like people just use compressors because it makes shit sound louder and in doing that you’re actually flattening out the music. I’m using a lot less compression on individual instruments than I ever used to. I can’t tell if that’s a factor of my ears changing or what people are sending me is more compressed than it has been. AT: So your approach with synths is not to compress them, but to widen them?
Martin Doherty on one of the mainstays of Chvrches sound, the Roland Juno 106
correctly receiving the intent of the artist. So the goal is to make sure that intent is coming across. Certain pieces of outboard gear help that intent come across. JUST TAPE OVER IT
AT: Do you really tape over the meters to placate your clients? RC: Yes, that’s true. I think those were actually the meters on my mix bus, the level coming out of the desk. AT: Do you drive things into that territory as a matter of course? RC: It depends on the music. That sound, where it’s not about headroom and it’s just about drive, works for certain pieces of music and not for others. The first time I did it I was mixing a Foo Fighters record. Dave [Grohl] had made a few records at that point and he noticed the pegging meters, so I started putting tape over everything. AT: To make him feel more at ease? RC: Yeah. Then I started drawing in meters where they probably should be, so he’d stop worrying about it. AT: Is that just more a function of your gear and how it performs. Or would you recommend or not recommend people trying it out? RC: I don’t think I could recommend anybody doing anything, but for a long time my meters would be pinning on a console, and I kept thinking it was because I was terrible and didn’t know what I was doing. Then I would pull the faders down and start again, and every single time AT 22
I’d end up right back where I’d started. And after a while I assumed I must just like the way it sounds. But it doesn’t work for everything. Some songs need something that has a bit more detail to it. But if the goal is to sound more arse-kicking, then often that helps. ELECTRONIC ELEMENTS
AT: You mix a lot of rock records, but you obviously have a way with electronic music, specifically the interaction between rhythmic elements and low end. Is there something special you aim to achieve down there? RC: Compared to a traditional rock album, that’s the whole key to the song: how the bottom end is moving you. And you usually spend a good deal of time getting that right. Having mix bus compression has a big impact on how that feels. How hard you’re hitting it and where the recovery time is set can affect the track if it’s pulsing or reacting to the bottom end elements. Sometimes if you have a lot of action in the kick drum but you’ve got a lot of sub synth in at the same time, then you usually have to put in a high-pass filter on the detector circuit of your compressor so that it’s not getting whacked every time a sub synth drops in. AT: So it’s more to do with how it pulses rather than simply how loud it is in the mix? RC: It’s how it mashes together. I can say I don’t usually compress the keyboards. If it’s a bass synth I wouldn’t really compress that too much, if ever. It will get compressed through a mix bus compressor and as part of a group, but if you start
RC: If it’s a major synth riff I wouldn’t want to use a widener plug-in because it starts throwing the signal out of phase and you’ll lose impact. But if it’s more of a textural element or a dreamy element, wideners help things stand out and generate their own space. I use Nugen Audio’s Stereoizer plug-in quite a lot, and occasionally a Brainworks plug-in, on synths. PICK OF THE CROP
MD: Why do you think Chvrches picked you to mix their debut? Having never worked with you, it couldn’t have been working methods or a relational bond. RC: I might have some kind of sonic imprint, but it’s not on purpose. In other words, there are things I like, but I don’t think other things I didn’t do are wrong. Everything is driven by the music that’s coming in. I’m pretty tough on myself and always trying to improve on whatever it is I’m doing, and will stop at nothing to make an improvement. I’ve
wanted to sit behind a pair of speakers and listen to music all day since I was five. That’s all I ever wanted to do. I was Philip Glass’s engineer for a few years and learned a lot of technical things. Rick Rubin liked the sound of a couple of records I’d done, so for a while I mixed records for him. That was interesting because he does not listen intellectually at all, only from his gut and instinct. Previous to that I listened almost entirely intellectually. He’s an incredibly intellectual person, but when he’d listen to the music he’d literally just listen to it like a teenager. The goal is to make sure it’s hitting you in that way, because when other people hear the mix or the song for the first time, they’re going to listen to it from instinct. The goal is to get all the technical shit out of the way so you can only work from instinct.
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FEATURE
Shure’s SM58 has been dropped from great heights, but its sales haven’t. It’s the real stuff of legends. Story: Mark Davie
One of Ernie Seeler’s original lab notebooks from 1957 depicting the Unidyne III concept. Every page was signed by hand and witnessed by his manager at the time. Seeler was also responsible for designing Shure’s SM81 condenser microphone.
McDonald’s has its special sauce, KFC the secret herbs and spices… Shure has a pneumatic shockmount. It’s not as saucy as other company secrets but, clad in a clandestine layer of 228 impenetrable mathematical equations, it has turned the SM58 into not just a cult piece of gear, but a ubiquitous one. The nature of its ubiquity is more than anecdotal, the Professional Audio Manufacturers Alliance places Shure as the number one seller of microphones in North America. Not in small part due to the success of the SM58. Putting the success of a microphone down to a shockmount that’s both difficult to understand and hidden from view doesn’t quite sum up the path to ubiquity. So let’s start at the beginning. AT 24
The genesis of the SM58 began well before people started singing into their ice cream cones. In 1938, 13 years after Sidney Shure first started shipping radio kits around the country, the company filed a patent for the world’s first single cartridge unidirectional, dynamic microphone. It was Benjamin Baumzweiger (later changing his name to Bauer), the chief engineer at Shure, who developed the design. Up to that point every unidirectional microphone design (think any microphone that is preferential to sound coming from one ‘uni’ direction) had been achieved by using two transducers. This original principle is still in use today in all multi-pattern condenser microphones, using the sum and difference of the two transducers to create all your favourite heart or number-shaped polar patterns.
TWO FOR ONE
But why use two, if you can do the same thing with one? Fewer unique parts is a catch cry of the manufacturing industry. Not just because it typically costs less to manufacture, but because, as Chad Wiggins, Shure Wired Systems Category Director, put it, “It improves consistency. Each part has its own unique tolerance, you stack up tolerances on 12 parts, you’re going to get more variability than if you stack up tolerances on five.” So Baumzweiger invented the Unidyne cartridge, one capsule to conquer them all — US Patent No 2,237,298. The essence of the Unidyne design is relatively simple. I say, relatively, because the concept is easy to get your head around, but the execution is not.
Single capsule microphones are all omnidirectional by default. Sound waves will curve and bend from every direction to get to the diaphragm. So even waves originating from behind a capsule will bend right round to the front of the diaphragm and shake things up a little. The Unidyne capsule proposed a new solution. If sound waves coming from behind the diaphragm were going to hit the front of it no matter what you do, you need something to cancel them out. So Baumzweiger proposed a resistance network that would take part of the sound coming from the rear and delay it in time so it would arrive at the back of the diaphragm at exactly the same time the rest of the sound would hit the front. This would effectively cancel out those unwanted sounds coming from the rear. It’s a simple concept that has been repeated in every dynamic microphone since its debut in Shure’s Model 55 microphone, including the SM58. UP TO MILSPEC
Soon after the Unidyne’s debut, World War II came along, and Shure started using its expertise to support the war effort. Throat and headset microphones for aviators, microphones for oxygen masks, battle announcement mics for the military were all developed in a couple of years but have helped shape Shure’s direction ever since. The company had to adhere to the military’s strict MILSPEC standards of reliability, but since he’d spent money buying all the test equipment anyway, Sidney adopted the military standards as Shure’s own from then on. Ever since then, people have been dropping Shure microphones from great heights, rolling over them in Mack trucks, and trying to drown them. There are plenty of SM series urban legends. In 1976, the world’s smallest dynamic microphone was introduced — the SM11 — and NASA duly stuck it to the outside of its spacecraft and sent it up on 25 missions. It eventually stopped working, and when NASA brought it in for a checkup the fault was in the cable, not the microphone. In the wake of Hurricane Katrina an SM57 that had been sitting in flood waters for two weeks was found as the waters receded. After a towel rub and some time in the sun it worked just fine.
microns thick, a dynamic is about 20 times that, to be able to mould it into shape. Still ridiculously thin. They call it a ‘micro’-phone for a reason. Even smaller is the amount the diaphragm moves under acoustic response. It’s imperceptible, even to a high-speed camera. Only by laser scanning the surface of a diaphragm can you pick up these tenth-of-a-micron movements. Whack the handle with a mallet though, and you can see the diaphragm move. So handling noise has far greater potential to disrupt the diaphragm than sound does. See the problem? When you hold a handheld dynamic microphone like the SM58 upright so the ball is on top, the diaphragm moves up and down in response to sound. Side to side movements have far less impact on the sound, maybe if you go fast enough you might induce wind noise. But the diaphragm moves up and down to create voltage, so that’s the plane of concern. We’re also not necessarily talking about big movements. Remember, the diaphragm of a microphone responds to tiny pressure differences. So rather than responding to being swept through the air, handling noise is made up of all those vibrations your hand transmits to the diaphragm by moving your fingers, clapping or thumping your hands against the microphone casing. Obviously foam and rubber can do a pretty good job at isolating a lot of things, but trying to calculate the expected load to isolate a studio is hard enough let alone calculating it for something that’s getting handled. Can you imagine trying to isolate a studio located in a tumble dryer?
“
Ernie Seeler was an absolute genius… so meticulous about his craft you can track his bathroom breaks by the rare 10-minute gaps in his lab notebooks
”
In Seeler’s Unidyne III patent drawings it shows the basic shape moulded into Shure’s dynamic microphone diaphragms to avoid breakup, enhance rigidity and strength, and provide better voice coil support. It’s been modified over the years but Seeler’s idea of moulding the diaphragm to increase rigidity and improve performance remains.
Whether you believe these tales or not is probably proportional to the number of times you’ve dropped an SM58 on the ground and familiarised yourself with its indestructible composition firsthand. Even the President of the United States will only use an SM57, well, two of them to be precise. There’s just no other mic the White House trusts more, having used a pair to deliver every presidential address for the last 40 years. There’s even a special SM57VIP dual microphone kit if you had the urge to double up just like the Prez. THE RUBBER HITS THE ROAD
Durability has contributed to the legend of the SM58. But here’s where the rubber, or shock absorbing material of choice, hits the road. A standard condenser diaphragm is about 2.5 AT 25
SHOCK TACTICS
Most microphone manufacturers rely on shock absorbing material to reduce handling noise. For instance, Heil’s PR22 handheld, which made a big buzz about its Iso Band internal suspension system (US Patent No 8,189,842), still relies on a Sorbothane rubber cup to reduce vibration. No one has surpassed the efficiency of the pneumatic shockmount Shure engineer Ernie Seeler developed with the Unidyne III capsule in 1959 (US Patent Nos. 3,132,713 and 3,240,883).
Seeler’s drawings of the full Unidyne III shows how integrated the mics are. Rest assured, ripoffs don’t do the math.
Ernie Seeler was an absolute genius. One of those rare characters so meticulous about his craft you can track his bathroom breaks by the rare 10-minute gaps in his lab notebooks. In a time well before computer simulators, even calculators, Seeler took three to four years with a slide rule, pen and paper to come up with the 228 equations an engineer would need to design a working pneumatic shockmount. Shure still has all his notebooks, hundreds of them, each documenting a few months of his tenure, locked away in banker’s boxes. “We’ve had a couple of acoustical engineers over the past few years who’ve gone deep into the books and vicariously relearnt the original science through these old lab manuals,” said Chad Wiggins.
Where it all started. The original Unidyne capsule in the Shure Model 55.
CAPPING RESONANCE A graph showing how dynamic microphones are tuned for presence peaks with a resonator cap, just like the SM58. John Born: “The resonator cap is like a Helmholtz resonator. A layer with holes perforated in it, which sits just below the foam, and above a cavity of air between it and the diaphragm. The cavity, and the diameter of the holes are tuned to a particular frequency range. More air and bigger holes, the resonant frequency goes down, less air and smaller holes, the frequency goes up. Where the actual microphone’s frequency response starts trailing off, the resonator cap boosts the top end to push out the frequency response.
Chad Wiggins: The presence peak, the 6k and 10k bump in the SM58 results from that resonator cap.
The original science is based around the idea that beneath the diaphragm and voice coil it’s glued to, there would be two chambers of air. They would act like cushions with a specific amount of resistance in between. The amount of resistance defines how fast air can flow between the two chambers: too much and the chambers would just decompress and clunk together; too little and it would be too rigid. But if you calculate the volumes and resistance just right, you get a perfectly functioning piston that absorbs most handling noise. And it works a treat. So meticulous was Seeler, that even half a century on, “The pneumatic math hasn’t really changed,” said John Born, Associate Product Manager for wired microphones. “The formulas haven’t changed, the values may adjust slightly to do a part change.” Even when the Beta 58 capsule was designed, the principles of the shockmount stayed the same, just the values changed. Born: “You can’t develop a perfect supercardioid capsule and then simply put the pneumatic shockmount on it. Because the second you vent to create that other cavity, you’re affecting the acoustic response. So you have to tune and make the pneumatic shockmount as you’re tuning and making the cartridge because they’re integrated and inter-related.” Although microphone manufacturers the world over have used the original Unidyne concept to create their own dynamic mics, and you could probably drop a bunch on the ground and have them survive, none can match Seeler’s pneumatic shockmount for dealing with handling noise. He’s the real legend behind the SM58.
AT 26
AT 27
TUTORIAL
PART III
STRIPPING THE CHANNEL Recreating traditional analogue paths is easy in-the-box, you’ll be surprised at what you can do. Tutorial: Dax Liniere
Let’s take a moment to think outside the box about the big-budget productions we’ve grown up loving. Historically, most have been recorded through an analogue console to multitrack tape, then mixed down through an analogue console to two-track tape. The choices of console and tape type are often different for each role. One of the accepted classic combinations is tracking through an old Neve (for the harmonics generated by their transformer-based designs), recording to two-inch Ampex/Quantegy 456 or 499 tape stock then mixing on an SSL (for midrange focus) to ½-inch tape. Signals travel from the console input preamps, through EQ, plus any outboard equipment used, to tape. This process is repeated during the analogue mix process — with the addition that it’s summed at the console master bus — for a double dose of analogue saturation. But by comparison, a ‘digital-centric’ production is
AT 28
likely to miss several of these key opportunities to impart saturation and colouration. These days, you can recreate this path with plugins for a fraction of the cost of even a single piece of hardware. Especially when you compare the purchase and maintenance costs of a console, tape machine and tape stock to inserting multiple instances of a plug-in you paid for just once. This affordability means you can own more than one virtual console or virtual tape machine, and it’s a great way to bring variety to your colour palette. CONSOLE EMULATION
First, let’s set about recreating the analogue signal path. You have two options: As discussed in the last issue, you can get the majority of your colouration via master bus processing, or you can use saturators on every channel. The second approach is more faithful to mixing Outside-TheBox (which is not to say it is more effective) and there are several plug-ins which will lend colour to your mixes.
Mellowmuse CS1V (US$79) VST/AU/RTAS www.mellowmuse.com/CS1V CS1V has two modes: A and B. The former is a vintage mode and the latter is a more open, modern mode. I found that the vintage mode was too closed for my liking, though I could definitely see it working well on more mellow genres like folk or even soul and rock ‘n’ roll. The modern B mode adds quite a bit of top-end sparkle which, if used judiciously, is a lovely touch. I also found CS1V to be one of the more dynamic sounding options.
THE SPICE OF LIFE
Console emulation is not the only reason you should consider using saturation on channels. Sometimes our source material is cosmetically lacklustre and needs a dab of makeup, or perhaps we want to go the other direction and ugly it up a little.
Terry West Saturn (Free) VST www.terrywest.nl Not to be confused with FabFilter’s flexible saturator of the same name, this plug-in packs a serious punch, adding quite a bit of liveliness and upper-mid character to a mix. The ‘US Pre’ mode adds a bit more ‘oomph’ and some brightness around 6kHz. Remarkably, this plug-in just happens to be free, but don’t let that scare you off, it sounds ‘like a bought one.’
Slate Digital Virtual Console Collection (US$199) VST/AU/RTAS www.slatedigital.com/products/vcc With their bus and channel plug-in set VCC, Slate Digital ambitiously sets out to capture the essence of five classic consoles. The Trident mode adds a nice widening effect to the low-mids, making it well suited to doubletracked rhythm guitars. The Brit4K mode (SSL) is one of the most open-sounding in the collection, but also imparts the least colour. While VCC’s US A mode (API) is tonally my favourite, its dynamics sound restricted compared to the other modes.
To me, the best thing about ITB mixing is not the cost, but the flexibility. You may have already realised that different console emulation plug-ins could be used in different places in your mix. In the box, you are free from the restrictions that working in the analogue domain imposes. You’re free to use the forward midrange sound of an API for your drums, the fatness of a Neve for your bass and the wide low-mids of a Trident for your guitars. Even the use of different tape types and speeds can be decided on a per-track basis. You cannot practically do this when mixing in the analogue domain.
Waves NLS ($249) VST/AU/RTAS/AAX www.waves.com/plugins/nls-non-linear-summer
Sonimus Satson (US$39) VST/AU/RTAS www.sonimus.com/products/satson Satson is a very subtle creature. It adds a slight thickness and widens the upper-mids. This plug-in also features handy high- and low-pass filters, plus a FAT mode to increase the drive.
Klanghelm SDRR (€22) VST/AU/RTAS klanghelm.com/SDRR SDRR was mentioned last issue as an excellent master bus saturator, and while it’s not specifically touted as a console emulator, the four modes (including Desk and Tube) make it equally well suited to channel applications.
For harsh or strident sounds — this could be cymbals, distorted guitars, vocals, brass, strings; almost anything, really — instead of reaching for an EQ to cut top-end, try a ‘warming’ saturator to round out the sound. A source that’s lacking bass or low-mids? Try a ‘fattening’ saturator like u-he Satin or FabFilter Saturn (the Warm Tape mode is a great place to start). Sometimes you’ll get a sound that’s too dull, so give it a bit of excitement with a saturator that generates strong upper harmonics. Maybe you have a loop or sample that’s boring and lifeless, if so, you could give it a bit of crunch. There are lots of saturators that can be used as a special effect. Try them on a small part you would usually tuck away in your mix. Like a sprinkle of chilli, it might just be the spice your mix needs. Plug-ins such as Voxengo’s Tube Amp, Togu Audio Line’s TAL-Tube and Camel Audio’s CamelCrusher are all very effective and also free.
Waves’ contribution to the market offers three consoles, each boasting 32 independently modelled channels. The mode modelled on Yoad Nevo’s Neve 5116 adds energy around 800Hz, whereas Mike Hedges’ EMI TG12345 Mk IV adds energy slightly above 1kHz. Both contribute a nice character to the mix that suits electric guitars and drums. They have a lot of similarities, but are different enough that you’d want to choose the flavour that best suits the song. Mike’s console also adds quite a bit of heft below 50Hz as it saturates on low frequency transients, a trait I would have expected more from the Neve. I found the emulation of Spike’s SSL 4000G was too easy to overload and, for me, didn’t suit a lot of sources. All in all, Waves’ NLS seems to be one of the more dynamic emulations in the pack.
VERDICT
My picks are Waves NLS Nevo for its depth, Mellowmuse CS1V for its great dynamics and Terry West Saturn for the price, plus an honourable mention to Klanghelm SDRR for best bang-for-buck.
Better still, you can choose to have no colouration at all on a sensitive source like vocals. This, ladies and gentlemen, is the single best thing about mixing In-The-Box: wherever you don’t want colouration, you don’t need to have it. While we can simulate the non-linearities of the analogue world in digital, analogue cannot pass a signal from one end of the chain to the other without imparting colouration and distortion. In short, InThe-Box can deliver the Outside-The-Box sound, but OTB cannot deliver the ITB sound.
DEVIL’S ADVOCATE
Part of the trick to a good sounding mix is uniformity. It’s all good and well to be able to choose a different console and tape type for every channel, but will that provide the cohesion you want? While logic and reason [not Logic and Reason, haha - Ed] can inform which colours you’ll use where, you must listen to how the final result sounds. Do you like it? Does it feel good? Because, in the end, listeners evaluate a song based on their feelings before their logical brain even has time to switch on. AT 29
MORE WIDTH
EQUALISERS
In Issue 96, I wrote about width in analogue mixes and how crosstalk affects the stereo image. There are many things at play inside any analogue circuit that may affect the perception of width, making it either wider or narrower. An analogue console contains thousands of electronic components: resistors, capacitors, inductors, transistors and integrated circuits. Although each of these has a specified value within the circuit design, the actual value may vary within a few percent, this is called the component’s tolerance. These minute differences between components accumulate throughout the signal path, resulting in subtle differences in frequency response, noise (amount and spectrum), dynamics and saturation. It’s been suggested that the noise present at a console’s output is perceived by the brain as an increase in reverb and because this noise is left-right uncorrelated (i.e., different on both sides) the ‘additional reverb’ appears wide. All of these leftright differences can contribute to the perception of subtle widening.
We’ve looked at the console input stage, its output bus and the tape it records to, but sitting in between all that is the humble equaliser, another great source of colouration. Every EQ circuit design has different properties that reflect the philosophies of its designer. Some are clean, some add character, some are surgical, and some ‘musical’. Whether Pultec, Neve, API, Helios or Trident, supplementing console EQs with outboard is a long-standing practice which can avail a broad palette of sound-shaping tools. New toys are fun, but learning the tools you have, especially your ears, is most important. The best advice I could give any engineer about equalisation is to practice ear training. The faster and more easily you can identify frequencies, the more efficient and better your work will become. This is especially true for live sound, where reaction time can be the difference between smiling and frowning musicians. It will also help you to identify how other mixes are constructed and how sounds are placed, which is a great way to learn and improve your own mixing.
The important thing to remember is that this widening is uncontrolled. Your stereo field (including your centre image) will always be slightly spread by an amount that varies with temperature and age. In the box, you can choose to apply plug-ins that simulate this behaviour only on the channels you want to affect.
TALKING TAPE
Variety Of Sound Thrillseeker XTC (Free) Win VST varietyofsound.wordpress.com Winner of KVR’s 2012 Developer Challenge, this EQ is a bit different from the norm. It features a three-band parallel-topology EQ that can impart quite a lot of character. Saturation is not added pre or post EQ, it’s actually part of it, with each band featuring Bootsy’s Stateful Saturation algorithms.
There are many ear-training apps available based around one-third-octave bands. Quiztones is one that’s available for iOS and Android and costs less than a beer. Now what are you going to do with these newly learned skills? Put them to good use with these great EQ plug-ins!
TAPE SPEED In the studio, common analogue tape speeds are 15 and 30 inches per second (ips). 15ips yields more noise, but also more character due to increased distortion and bottom-end, and subdued top-end. 30ips is generally cleaner and preserves more top end, but has less bass ‘fatness’.
UAD-2 API Vision Channel Strip (US$299) UAD-2 www.uaudio.com/store/channel-strips/api-visionchannel-strip Punch and character sum up this plug-in. Going well beyond the recently released bundle, the API Vision plug-in is a full channel strip that models their 2520 op-amp, largely responsible for the signature API sound. For those with an analogue console background, this channel strip might be for you. The 550L module offers four bands of EQ while the 215L takes care of low- and high-pass filtering. It also features the fantastic 225L compressor and 235L gate modules.
TRACK WIDTH Common tape width/track counts include 24 tracks on 2-inch, 16 tracks on 1-inch, and two tracks on ½- and ¼-inch. As track width increases, you get increased dynamic range (due to a lower noise floor) and decreased cross-talk. TAPE FORMULATION Common tape types are Ampex 456, 499 and GP9. 456 is a low-flux tape which saturates more easily and hence, has more character. GP9, takes quite hot levels and has a ‘modern’ clean tape sound. 499 is approximately half-way between 456 and GP9. Though Ampex has been out of business for a while, and the Quantegy revival is slow, if you’re serious about recording to tape, you’ll need to check out ATR or RMGI for equivalents. MACHINE SET UP To get optimal performance from any equipment, it must be set up correctly. Different tape types have different properties and this requires the machine to be calibrated according to the tape manufacturer’s specifications to perform as promised. Bias is a function that offsets the signal fed to the recording head by a certain amount and it’s designed to maximise sensitivity and provide a ‘clean slate’ for magnetisation. However, bias also affects frequency response and distortion. When bias is set as per the tape manufacturer’s specifications, it strikes an even balance between frequency response vs distortion. Under-biasing will result in more top-end, but also more distortion. Over-biasing, while reducing distortion and delivering a cleaner bottom-end, can also subdue the top-end. AT 30
UAD-2 Pultec Passive EQ Collection (US$299) UAD-2 www.uaudio.com/store/equalizers/pultec-passive-eqcollection
Variety Of Sound BootEQ MkII (Free) Win VST varietyofsound.wordpress.com This is a very musical-sounding implementation of a digital EQ with analogue-shaped curves, presented in a Lunchbox-inspired GUI. BootEQ also features a preamp simulator with variable drive and additional tube simulation to introduce pleasant second-order harmonics.
A revamp of the 11-year old Pultec EQP-1A plug-in, the new collection, also containing the MEQ-5 midrange EQ and HLF-3C filter set, is a vast improvement. Inserting this plug-in brings an instant, pleasant change to the sound as harmonic structures from the original tube hardware are emulated. The new EQP-1A has a very ‘tubey’, warm and gluey sound to it, making it perfectly suited to a variety of sources.
DEVELOPER INTERVIEW: Klanghelm’s Tony Frenzel
“
In short, In-The-Box can deliver the Outside-The-Box sound, but OTB cannot deliver the ITB sound
”
DL: What is your approach to making an audio processor? TF: I’ll go back to a time when I called myself a musician. I used to always blame the tools, which is very bad for a musician, but it’s very good when you start coding plug-ins. It made me very sensitive and picky about tools, how they have to work. I have a special sound aesthetic in my head derived from favourite producers, bands and records. I want my plug-ins to become part of a sound, not like a coat you put on. I see the user interface as a very big part of development and I put a lot of thought into them. To me, it’s like music and lyrics; sound and user interface have to work together. After a few weeks of coding a plug-in, once I start to get an idea of how it could sound in the end, I start making a UI that reflects that. Can you work fast, can you see every control you need? The UI has to be slick, fast and enjoyable to look at.
TIP: SECRET TO TIGHT BASS
To tighten the bottom-end of your mix, you need to think beyond the obvious sources of bass (such as kick drum, bass guitar and bass synths), and turn your attention to the rest of your tracks. Unwanted sub-audible energy eats up our dynamic range without any benefit and is a common cause of a muddy mix. Not only affecting headroom, these infrasonics can combine with the useful bottom-end of bassy sources, producing an uneven frequency response due to constructive and destructive interference. Most hardware consoles have high-pass filter (HPF) buttons on every channel, but most DAWs do not, even though they are most important to an ITB project. Consider which instruments have bottomend that you want and which are causing clutter in your mix, then put an EQ plug-in in the first insert slot. Don’t forget to set the crossover frequency of the HPF to suit the source. Console HPFs are often fixed at 100Hz and this is a good place to start in-the-box, though don’t hesitate to drop it to as low as 30Hz or as high as is appropriate for the mix. 180Hz is not uncommon for certain instruments within the context of a mix and I have high-passed hi-hats at 400Hz in some cases. If your DAW allows it, save a default signal path for each channel with a bypassed EQ in the first insert slot. Only enable it when you need it. Speaking of bass, try the old simultaneous boostand-cut trick with the LF shelves of any Pultec. This seemingly illogical technique works because the response curve of the two shelving filters is slightly different, producing a bell-shaped dip just above the shelving frequency. Try it for some serious thump and rumble. With PSP NobleQ, you can adjust this frequency difference to tailor the sound.
Dax Liniere: When did you start? Tony Frenzel: In the ’90s, when I built a guitar. I found it was much more fun to work with the electronics than building the guitar with wood. Since I started coding in the ’80s, I never knew what to do with my skills. I was never interested in coding or playing games, I found that boring. I always wanted to do something with music. All my life I’ve only had music in my head. It was very exciting for me when Steinberg released VST. It gave people the ability to build plug-ins, virtual gear, with just a few lines of code.
PSP NobleQ (US$69) VST/AU/RTAS www.pspaudioware.com/plugins/equalizers/psp_nobleq NobleQ from Polish company PSP Audioware is a great Pultec emulation alternative for non-UAD-2 users. PSP doesn’t claim it’s an exact replica, instead they’ve recreated the Pultec’s essence, while adding features not found on the original including inbetween frequency positions, variable tube warmth and 30/40kHz shelf frequencies for adding ‘air’.
FabFilter Pro-Q (€149) VST/AU/RTAS/AAX www.fabfilter.com/products/pro-q-equalizer-plug-in This EQ plug-in is my personal go-to. Pro-Q’s clean and intuitive interface is a perfect reflection of its sound and usability. With 12 filter shapes and up to 24 bands, it covers traditional and linear-phase equalisation in one plug-in, with split-stereo and M/S modes plus a spectrum analyser for hunting down narrow troublesome resonances. If I had to make a record with only one plug-in, it would be Pro-Q.
DL: What did you set out to achieve with SDRR? TF: It actually started as an EQ, but I found working on the input and output saturation more rewarding. I didn’t want to reproduce something that already existed in analogue or other plug-ins. I didn’t care if it sounded ‘analogue’ or not, that wasn’t my goal. There are so many saturation colours that haven’t been done yet, like a digital saturation that sounds pleasing and unobtrusive. As a guitarist and collector, I always found old fuzz pedals very interesting. Electronically they’re very primitive, but the sound is very complex. I wanted to make this warm, complex sound suitable for mixing, create it with a fairly flat frequency response for subtle work. DL: I heard that you were recently able to quit your day-job to concentrate on Klanghelm. TF: It was about time! (laughs) I worked on SDRR for over a year. Having two jobs for the last two years has been quite exhausting. You can’t code plug-ins in your free time because it makes circles around your head; you can’t think about anything else. I just hope I can make a moderate living out of it for my family. I guess nobody in the plug-in business can be rich. But it’s very rewarding, especially when you know some of the bigger names are using your plug-ins on records. You listen to the radio and know that a little part of you is in this record. That’s really a great feeling. I hope it will work with these low prices; I don’t want my plug-ins to be seen as an investment, I want them to be used.
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FEATURE
What happens when an orchestra sends its virtual clone on the road and lets punters remix its performance? Story: Mark Davie
The big money is on the road. It’s what everyone’s saying. That albums are just an audio press release for the tour and the t-shirt. But night after night of playing the same repertoire can be a drag. And you can only ever be in one place at a time, severely limiting earning potential. If only there was a way you could multiply yourself, be in more than one place at a time. The Australian Chamber Orchestra (ACO) and Mod Productions have gone one better. They’ve figured out how to play in multiple places at once, without physically being in any one of them.
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PITCH HITTER
The ACO first got in contact with Michela Ledwidge, Director of Mod Productions, after she gave a presentation at an ABC event on her history with re-mixable film and interactive story-telling. It hit the nail on the head for the ACO, which had been researching ways it could expand its audience. Michela: “I was commissioned to spend a week looking to see if there was an opportunity to leverage the kind of interactive film projects I’d done in the past. Our framework was all about how to get the audience involved in a high-end
audiovisual experience where you don’t have to interact, but if you do interact, there’s a real remixable potential.” There were a number of criteria the solution had to satisfy. The obvious one was broadening the reach of an orchestra that can only ever be in one place at one time. But also, the costs of touring an ensemble are enormous, so finding a more costeffective distribution of its performances was also high up the list. But lastly, the ACO’s audience is generally longer in the tooth, so if there was a means of bringing younger audiences into play, that wouldn’t be sniffed at either.
The result of all the research is the ACO Virtual roadshow. A travelling kit of seven video projectors, an 8.2 (minimum) surround speaker system, two spec’ed out PCs and an iPad on a stand that lets attendees shine the virtual spotlight on specific performers as well as giving detail about the four-song repertoire from Bach, Grieg, Smalley and Piazzolla. The numbers are promising. A typical ACO 10-day tour might ring up costs in excess of $100,000, whereas Michela estimates the total cost of putting together the open-ended ACO Virtual at about $400,000. Mod Productions absorbed a lot of the labour costs because they now own the technological intellectual property. It means Mod can use the staging format for other similar shows in the future. So far, ACO Virtual has performed at the Gold Coast Regional Arts Centre and Swan Hill Gallery. But the plan is to keep the touring package out on the road for at least a couple of years, with plenty more venues showing interest in staging the concept. To stage the show, the venue hires the kit off the ACO. And the ACO production crew has got it down to a one day bump-in, and three-hour bump out. If it keeps going well, and funding permits, a second kit might be pressed into action, and more repertoire recorded to keep ACO Virtual ticking over for years to come. UNDER THE SPOTLIGHT
When ACO Virtual rolls into town, the performance is projected on the walls of a room with the speaker system distributed between each projection. The trick is being able to precisely overlay the phantom audio image over each player. Michela explains how the system integrates with the projections on the gallery walls: “In the centre of the space is a music stand
with an embedded tablet. As each piece starts, a full body image of each player in that piece appears on the tablet screen. By swiping your finger across the screen you toggle, or select one or more of the players to be spot-lit on the walls. “When a player is spot-lit, their lighting comes up on the main displays and the audio is remixed automatically so the audio from the spot-lit players is brought to the fore. It’s quite a subtle effect because it’s not simply turning the volume from the video up to 100%. It takes into account different weightings depending on what instruments you’re selecting and the number of players. It’s quite a sophisticated, interactive audio patch that is doing the dynamic changes.” It’s quite a big risk, putting the control of your mix in the hands of laypeople, but Michela has a vision for the future, and re-mixable performances are a big part of growing interaction with audiences. Michela: “The ACO is an adventurous bunch that took a leap of faith to allow its recordings to be remixed by punters. We’ve put parameters around what the remix experience is, but it’s exposing members of the ensemble in ways that they would never otherwise be exposed. So there’s still a degree of bravery required to enter this space and it’s not been the easiest vision to sell. I’m biased, but from my perspective, it’s definitely the next phase.” THE ACO IN A BOX
Michela: “The show is currently running off two very high spec PCs with RME MADI cards and two $2500 graphics cards. On the core engine is a program called TouchDesigner, which manages up to 39 different videos worth of material and triggers all the audio as required. We built layers of software on top of that for our logic, and the web management is all controlled and managed by a web service.
“
The ACO is an adventurous bunch that took a leap of faith to allow its recordings to be remixed by punters
”
“On the second machine we’ve got Plogue Bidule with a range of plug-ins for sweetening and acoustic modelling tuned for each venue. The show control messages come to Plogue Bidule via TouchDesigner, and TouchDesigner gets its messages from the audience via a custom-built iPad app. “There’s a huge amount of traffic going between the audio and video PCs. A MADI connection transfers all the audio, and some additional logic inside Plogue manages all the OSC messages coming from the audience interface to figure out how a particular mix can occur without drowning out the violins with the bass, for instance.” Simon Lear, the ACO’s sound designer for the project picks up the trail: “Plogue Bidule is like Max/MSP but higher level. If you want an eight-channel mixer you don’t have to build it from scratch. Just grab a device and drop it. It’s a visual, modular audio-programming interface. It takes the MADI input, and based on the conditions it does various things with it. The real hub of the audio side is a plug-in from Flux called IRCAM Spat — a surround spatialisation and reverb processor that generates the AT 33
(left) By swiping your finger across the screen you toggle, or select one or more of the players to be spot-lit on the walls. When a player is spot-lit, their lighting comes up on the main displays and the audio is remixed automatically so the audio from the spot-lit players is brought to the fore.
localisation and the reverb in real time. It’s hosted in Bidule as a VST plug-in, and it comes out of the MADI card to an SSL D/A converter and goes straight to the speaker.” Each time the kit is rolled out in a different gallery or performance venue, the system has to be re-tuned to make sure the audio localisation matches the new positioning of the projections and the players within them. Simon: “The audio is a phantom image between the speakers — it’s a virtual position so we have to tune that localisation to each space and physical setup. At that point I hand over to sound technician, Felix Abrahams, that’s his department. He installs it and does a fantastic job of tuning the system and its localisation.” The automatic mix is not just a matter of turning one player up and the other down, it fluidly transitions between selections, as well as figuring out what to do when the orchestra gets switched back in. Simon: “If you’re going from a full orchestra to one player spotlight, there’s going to be a natural drop in level which could make it feel less impressive. So when you go from a full mix to a single person, you need a level boost, and then if you add people you still want a boost but less of a one for each player. There’s no compression or limiting going on, it’s just manipulating the levels based on conditions, and all the ambience is generated in real-time. As more people are added, it dynamically adapts — it’s like a matrix of gain settings.” SPLITTING UP THE BAND
The ability to manipulate individual recorded parts in an orchestra must set off warning bells for the classical engineers out there. It’s probably obvious by now, but the recording process didn’t AT 34
(above) The orchestra had to be captured individually for the system to work, and not only that, each performance had to be filmed against a green screen too.
follow any of the standard orchestral miking techniques — no Decca trees fanning out above the ensemble. The orchestra had to be captured individually for the system to work, and not only that, each performance had to be filmed against a green screen too.
“I knew from doing tests from a pilot with the quartet that I’d get about 20dB of separation between players. And that worked with the brief because the players didn’t want to be 100% isolated in the final installation and for some of the pieces that wouldn’t work at all anyhow.
It was a conundrum, because orchestral players aren’t used to operating in isolation, and using a click track would be a leap too far in the wrong direction. In the end, the orchestra was set up on a film sound stage. Each player was stationed on a plinth a couple of meters apart with the two main desks of violins and violas facing each other, while the cellos and bass rounded out the horseshoe configuration.
“There was a settling period working out how much isolation there would be, because different departments had different views. But it found its own way, and in the end the spill from the mics fell into the groove. It wasn’t too destructive.
Simon: “The sound stage was a nice, big, open room with quite a bit of absorption built into it, but it didn’t sound like a great hall or studio. Much more than that, it was a real challenge for the players to play physically separated from each other because they’re accustomed to standing next to each other.” The players weren’t the only ones in an unusual situation. Simon had to figure out how he was going to capture each instrument without sacrificing overall tone and dynamics. Simon: “I individually miked them with Schoeps MK4 small diaphragm cardioid condensers, and DPA 4061 bug mics on Shure radio packs, which we really didn’t end up using. They were a bit of an aid in post-production and I just wanted them for a backup. I had a few Beyer M160 hypercardioid ribbon mics on principals, that were more of a friendly frequency response I could use as a reference. But most of what you hear in the installation is just a straight single Schoeps per player.
“We hired a redundant recording system which was based on API 8MX2 preamps split into dual Tascam X48 [digital multi-track] recorders. And we used a Midas console as a monitoring hub.” Once the best take was chosen, Lear did a bit of post-production cleanup in Magix Sequoia — getting rid of noises, thumps and clicks — but didn’t use much EQ on the individual instruments. The main EQ’ing happens in set up to attain the right room balance. And because the balance could be mixed on the fly by punters, a lot of the effort was in setting rules for the automix logic. It’s a strange way to record an orchestra. But allowing punters to participate in a performance by spotlighting players and following along with the score has brought down the barriers for people unlikely to set foot inside a concert hall. But has it replaced the ACO’s touring schedule, is everyone now sitting at home without a job? Simon Lear says the orchestra is going “120%”. Better than ever. ACO Virtual is just another string to the bow, a way of doubling the coverage of an orchestra already spread thin. Now if only we could all do that.
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REGULARS
MAC NOTES What drives you to the brink? Column: Anthony Garvin
Back in 1999 when I started getting into all this, recording audio onto a computer hard disk was not the easiest task — IDE hard drives had seemingly just started catching up to our DAW’s demands and pricey SCSI options were the only other viable option (if you could afford the drives and the witchcraft-like method of getting them set up). Fast forward 14 years and it’s now a fairly straightforward task to record 100+ tracks of audio onto a USB 3.0 drive — or built-in SSD of any MacBook variety. After pouring over specs and building systems for many years (and reviewing the G-Dock EV Thunderbolt drive elsewhere in this issue) I’ve come to the realisation that our demands aren’t that ‘special’ anymore — and bandwidth is not really an issue like it once was. A 24-bit/96k WAV consumes 16.48MB of storage per minute. Divide that by 60 (seconds) and you get 0.275MB/s — a number which is comparable against the throughput of various protocols like USB, Firewire and Thunderbolt, and the devices that harness these. By doing a few basic mathematical divisions, we can find the theoretical maximum playback track counts for the protocols in the table below (using expected real-life throughput).
[Note that 1MB (Megabyte) = 8Mb(its), so if you are looking at the latter nomenclature, divide the figure by eight to get Megabytes.] Looking at these numbers, it’s interesting to note the maximum throughput of the protocol vs the speed of the storage device — for example, using an internal 7200RPM SATA drive in a current Mac Pro will only allow 171MB/s, despite the SATA (rev 2.0) spec allowing for 300MB/s (so we can only play back 621 24-bit/96k tracks instead of 1090!). Okay, so all of this is theoretical calculation and we do need to factor in the real world, where real-life tests don’t often live up to the specs for various technical considerations. Also bear in mind protocols like USB, Firewire and Thunderbolt can often run other devices on the same bus simultaneously, which reduces the bandwidth available to playback (or record) audio from a storage device (as it is sharing the bandwidth with the other devices). To bring all these numbers into the real world, there’s a nifty (and free) application available on the App Store made by Blackmagic called Disk Speed Test. (Blackmagic make video production and editing gear, where their hard drive demands are far more intense than for our DAWs). Using the application, I can test any storage device
PROTOCOL
MB/s
24-bit/96k Track Count
USB 1.1 (Full-bandwidth)
1.5 5
Firewire 400
50 181
USB 2.0
60 218
Firewire 800
100 363
SATA (rev 3.0)
600 2181
USB 3.0
625 2272
Thunderbolt 1
1250 4545
Thunderbolt 2
2500 9090
DEVICE 5400RPM Hard Disk (WD 2.5-inch Scorpio Blue WD10JPVT)
144 500
7200 RPM Hard Disk (WD 3.5-inch Black WD4003FZEX)
171 593
SSD Drive (Crucial M500 series)
500 1736
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of my liking to discover its practical limitations. For example, my LaCie Rugged portable drive (containing a WD Scorpio Blue 5200RPM 2.5-inch hard disk) shows a read and write rate around 51MB/s. The internal SSD drive in my 13-inch MacBook Pro Retina reads data at about 477MB/s and writes at about 397MB/s — that’s at least seven times faster. Bear in mind also, as my internal drive is running the Mac OS and looking after tasks like virtual memory, which require storage use, this calculation is understating the drive’s actual full bandwidth. You’ll also notice that the write rate is about 20% slower than the read-rate, which is usual for less-than-really-expensive SSD drives. To further test the practical implementation of how all these numbers stack up in actual DAW use, another application called iStat Menu (from Bjango, US$16) will display real-time hard drive read and write rates. Whilst playing back a 100-track 24-bit/96k project in Logic Pro X (version 10.0.3), it shows me that the read activity on the drive varies between 17 and 39MB/s. Presumably this variation as it plays is because Logic buffers the audio it plays back, meaning it takes the data it is about to play off the drive in chunks to play back as it needs, rather than going straight from the drive to your speakers in a steady stream. I tried this on a variety of Thunderbolt, USB 3.0 and internal hard drives — and found that they all played back the project without an issue. I could also punch-in 10 tracks to record at any point, all at a buffer size of 128 samples. So, feeling safe about my current and future DAW demands, there’s another factor to consider — data copy times — there’s nothing worse (nothing!) than waiting for a 100GB backup to finish copying to a Firewire 400 drive at the end of seven days of 12+ hour sessions. Whilst it appears that almost any old drive will suffice for working on 100+ track projects, bear in mind the speed of the protocol and storage device you are using for backups. A USB 3.0 7200RPM drive is probably a better option than a Firewire drive — as the quicker you can get a backup done, the sooner you’re out of the studio at the end of the day!
REGULARS
PC AUDIO The old adage ‘Waste not want not’ applies to PCs too! Column: Martin Walker
My first music PC had a 25MHz processor and 2MB RAM, so it’s perhaps understandable that I sigh when today’s musicians grumble about their incredibly powerful music computers running out of steam. However, there are various quick and easy ways to make the most of any PC’s resources, so please try some of the following if you seem to be running on empty. First up, if you want to squeeze the last drop of performance from your mechanical hard drives, they may still benefit from an occasional manual defragmentation (re-organising their files to minimise physical head movements and therefore speed up drive transfer rates). Having said this, most hard drives are already significantly faster than they were a few years ago, so defragging is unlikely to result in noticeable performance hikes unless your drives are already getting rather full or haven’t been defragged for many months. Meanwhile, the solid state drives now used by so many of us for speeding up Windows (and sometimes even our sample libraries) shouldn’t be defragmented at all, as they contain no moving parts and may even have their life span shortened as a result. Indeed, if your PC is running smoothly, yet still struggles to run your largest projects, modern hard drives are unlikely to be to blame, unless you’re attempting to run over 100 simultaneous audio tracks. And while adding more RAM can sometimes help if you suspect the lack of it may be the cause of your own problems, there’s never any need to guess, or to blindly upgrade just in case – just load in your most complex songs and monitor their RAM usage for yourself. You can read the Available Physical Memory in kilobytes from the Performance page of Windows Task Manager. PROJECT MANAGEMENT SKILLS
In my experience the most likely contributor to running out of steam comes down to CPU overloads, and most CPU ‘maxing out’ problems are simply a result of attempting to run more plug-ins or soft synths than your PC can handle. The most significant audio parameter relating to CPU consumption is sample rate, so although
most audio interfaces now offer 24-bit/96k and even 24-bit/192k options, reducing your song’s sample rate to 44.1k (still arguably the most sensible option) will more than halve your CPU load at a stroke, and is unlikely to make an audible difference unless you’re recording acoustic instruments with high quality gear. Bit depth isn’t linked to CPU, and 24-bit recordings shouldn’t eat up any more processor cycles than 16-bit ones, so my projects are still almost all running at 24-bit/44.1k. Another phenomenal waste of CPU can occur when modern musicians jump straight into writing music on a PC, bypassing the traditional learning curve of using analogue mixers and rack effects, and therefore fail to realise the huge benefits of the aux send. I’ve spotted beginners grumbling that they can only run eight software plug-in reverbs, which is a huge waste of CPU. It’s rare to require more than one or two different types of reverb per song, so instead of placing individual reverb plug-ins in the insert slot for each audio channel, patch them in as an aux send effect, and then you can add variable amounts to each and every track. This approach also works with echo, chorus, and any effect where the dry and wet sounds are mixed — only plug-ins like EQ and distortion, where the whole sound is treated require an insert. In many cases, it’s software synths that eat up the majority of our PC processing power, but you can often reclaim a lot of this with very little audible effect by capping polyphony. Many soft synths let you choose the maximum number of voices, and unless you’re into complex jazz chords, most pad sounds will be happy with 16 voices or less without note-robbing being noticeable, while guitar sounds shouldn’t need more than six voices, and bass and melody lines can be restricted to monophonic status. Even if your soft synth doesn’t provide such facilities, some MIDI+Audio sequencers have a ‘Restrict Polyphony’ function to edit your MIDI data to the same end. Finally, most audio interfaces exhibit a rise in CPU due to ASIO driver overheads below an audio buffer size of about 12ms. So if
you’re running out of CPU and using a smaller buffer than 12ms, you’ll reclaim processing power by increasing it. If you’re currently running at 3ms or below you may be able to reduce your CPU load by 50 percent or more! JUNK SOFTWARE – JUST SAY NO!
Ironically, another way to avoid your PC slowing down over time is to avoid installing junk software. Are you really going to continue using that ‘must have’ freeware utility or plug-in after the first few tries? If not, just say no. Also, be very careful when installing legitimate software not to blindly accept its offers to add other ‘bonus’ items by default. For instance, Adobe’s useful Flash Player will install McAfee’s Security Scan Plus, while Java updates will install the Ask toolbar. Over time, all these unnecessary choices and hidden extras will bloat your Windows install, clog your registry, increase your start-up times and may even impinge on audio performance by running background tasks at odd moments. To find and get rid of most unwanted software just use the Windows uninstall program option from Control Panel. To discover tasks that are currently increasing your start-up times, just use the Startup tab of Windows’ MSCONFIG utility to see what’s being loaded into your PC each time you boot up, and disable anything that’s not strictly necessary (how many routines really do need to check daily to see whether their associated application has been updated?). An even easier alternative is to use one of the few third-party utilities that still merits being downloaded. Piriform’s CCleaner (www.piriform.com) continues to be improved while remaining free in its basic incarnation. Apart from its comprehensive clean-up of temporary browser, system and application cache files it also includes a useful Registry Cleaner, a very similar Uninstaller to the Windows one, a better viewer/editor for Startup tasks than MSCONFIG, and a utility that searches out duplicate files (you may be surprised at just how many there are!) all in one handy 4.2MB package. Power to the PC! AT 37
REVIEW
NOT MESHING AROUND The AT5040 is a big mic with a modern look. It’s cylindrical with a non-reflective steelygrey body made from aluminium and brass. The head is made of two layers of mesh that provides physical and pop protection, but still let you see through to the elements.
MAGNETIC PULL The supplied shockmount is an eyegrabber. It’s a space-age design, a sort of metallic hand, where the mic pushes into the hand until it clicks to let you know the magnets, yes magnets, have grabbed the mic. A small locking handle can then be turned and the mic is secure. If you push it in and it doesn’t click the mic will hit the floor when you let it go. I prefer the security of shockmounts where the mic screws into the mount but this is pretty good once you know how it works… and it looks fantastic. It will impress the clients.
AUDIO-TECHNICA AT5040 Studio Vocal Microphone Audio-Technica’s flagship AT5040 has broken the large diaphragm condenser mould with four rectangular capsules that are greater than the sum of their parts.
NEED TO KNOW
Review: Mark Woods
PRICE $3999 CONTACT Technical Audio Group: (02) 9519 0900 or info@tag.com.au
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PROS Biggest LDC available Non-resonant, pure sound that brings the source forward Audio-Technica top build quality High output/low noise
CONS No HPF No Pad
SUMMARY Audio-Technica’s flagship mic is a triumph of technology. Breaking open the large diaphragm condenser sphere with a quad-capsule rectangular design. Its presence, heightened sensitivity, and extremely low self-noise brings vocals up to the front of the mix — right where they should be.
I’ve liked Audio-Technica mics ever since Depression’s singer Smeer gave me a pair of PRO4L dynamics as payment for a recording I did for the band. That was all the way back in 1987. They looked like they’d had a hard life at the hands of too many gobbing thrash-punk bands, but I cleaned them up and got years of use out of them as a general purpose vocal/tom/guitar mic. They always worked and sounded pretty good at the time. Currently, I don’t do many live shows, or recordings, without using at least one of Audio-Technica’s more recent products.
response problems. And speaking of phase, the AT5040 cleverly uses the four elements to create a type of balanced circuit to reduce any interference between the diaphragm and the electronics in the body of the mic. This helps it achieve a self-noise figure equivalent to 5dB SPL, i.e. it will take 5dB SPL to create the same voltage as the microphone’s own self-noise would produce. I don’t recall seeing a lower figure for a comparable mic, [Shure’s KSM44A LDC has 4dB SPL of self-noise, which is also a formidable number - Ed] but then again, there isn’t really a similar contender.
I’m particularly fond of the AE5400 vocal mic and the AE5100 instrument mic from its Artist Elite Series for live sound, while the classic AT4033 gets used regularly for both live and studio sound. It’s probably no coincidence that my current favourite models are all large-diaphragm condenser (LDC) designs. Audio-Technica is good at making LDC microphones. And it’s recently released the biggest of them all — the AT5040 Studio Vocal Microphone. It’s the first (and only) mic in the new 50 Series and it takes the LDC design into brave new territory.
LIVING LARGE
BIG AS BONES
Large-diaphragm condenser mics are the big boys of microphones; both physically and sonically. They offer several technical advantages over other mic designs including high output level, wide dynamic range and low self-noise… but it’s the character of the sound they produce that makes them the common first choice for the most important of recording tasks; vocals. Big lows, rich mids and smooth highs are common descriptions of the sonic character of LDC microphones, and who wouldn’t want all of those in their vocal sound. Bigger may be better but there is a limit to the size of the diaphragm. Around one-inch in diameter seems to be the limit before the diaphragm becomes unwieldy and anomalies appear, including loss of high-frequencies and reduced transient response. The shape of the diaphragm also affects their performance; most LDCs use circular diaphragms but these can create resonances that are hard to control. Rectangular diaphragms can reduce these resonances and while it’s traditionally been difficult to tension them accurately, it can be done, and there are currently just a handful of mics on the market that use them. Technically, Audio-Technica has really pulled one out of the hat with the AT5040. The diaphragm is constructed of not one but four optimally-sized and tensioned rectangular elements. These are placed in a block and synced together to act as one diaphragm with twice the surface area of a normal one-inch diaphragm. To achieve the same surface area with a circular diaphragm it would need to be 1.5-inches in diameter. Very clever. This feat of engineering relies on manufacturing the individual rectangular elements accurately enough so they can be combined without fighting each other. It’s easy to imagine that even tiny differences in the response of the individual elements would lead to phase or frequency
The day the AT5040 arrived I had a show at the Theatre Royal in Castlemaine and the support was my mate Archer Shepherd. Archer does a 1930s hobo sort of thing; solo, just voice and old acoustic guitar. He’s a regular performer at shows I do and he likes to use the single condenser mic technique popular with the bluegrass/americana scene. Archer knows I like to use him to test microphones. I know the AT5040 is designed as a studio mic but live shows are a great test of a microphones reach, colour, off-axis response and isolation. The first thing I noticed on firing it up was the combination of low noise and high output; for a moment I thought it was going to be too unstable, but quickly realised it was wide open and it was actually quite stable. It just needed much less gain than I expected and less low end than the amount that was flying at me — the bottom end of this mic is huge. Most condensers want to take off somewhere below 100Hz with big live PAs. PAs are often tuned for strong bass and the mics are pretty much omni-directional at low frequencies — the AT5040 was typical in wanting to take off around 50-60Hz in that room. It was easily fixed with a low-cut on the channel, which was all the EQ required. Once the gain and the low end were sorted I ran up the PA. It went up to a good level in the frontof-house without becoming unstable… half the battle with these sorts of mics at live shows. Then I began to enjoy the tone. Full low-mids, easylistening mids and transparent highs combined to create an immediately impressive sound. There was also something distinctive about the presence, particularly in the mid-range, that seemed to somehow magnify the sound. Archer sounded closer to the mic than he looked. During sound check we did notice it’s got a fairly narrow address angle, especially in the vertical plane. Because he uses a single mic the height of the mic changes the balance between guitar and voice, we have a ‘normal’ position but found the AT5040 was quick to lose level if he got even slightly above the diaphragm, more so than other mics we use. I ended up angling the mic slightly upwards to stop him bending down to find the sound. It’s usually not possible, or practical, to use high-end studio mics on live stages given the relatively risky nature of live shows. But when they work it can be the best and purest sound you’ll hear through a PA. The AT5040 is too expensive to use as a live mic but on this occasion it was a treat to hear the quality it had to offer on
Inside the AT5040 is a formidable piece of rectangular 4-in-1 diaphragm engineering.
the reasonably big stage. No-one spilt beer on it, and I left the show looking forward to trying it in the studio. SAFE AND SOUND
The studio is its natural domain, and much safer. I spent a few pleasurable weeks using it on everything I could and because my customers often play acoustic/folky stuff I do lots of vocals and acoustic instruments. In a similar way to the live show some preparation was required before launching into recording with the AT5040. For vocals, first you need a pop shield… and maybe another because although the mesh grille looks cool, and lets you see the elements of the diaphragm, plosives get through quite easily and they are a big event. Then there is the level; this thing puts out lots of it, easily enough to overload some of my preamps. The AT5040 delivers 56.2mV per pascal, which is very sensitive, and not really a surprise given its generous capture area. Combined with its ability to handle maximum SPL of 142dB SPL, you could have a possible output of around +26dBU. To give you an idea, maximum input level on an AMS Neve 1073DPA’s mic input is +6dBU. That said, you’re unlikely to hit those kinds of SPLs (you should be able to hit about 122dB SPL before overloading the Neve) and you’re not going to be shoving the AT5040 into a kick drum any time soon. But it’s something to keep in mind when choosing the right preamp to match the AT5040 — it’s hotter than most LDCs. So you may need an in-line pad… which makes you wonder why a mic with such a high output doesn’t have a pad built in. AT 39
Similarly, something to keep in mind is the bass response below 100Hz. It’s slightly boosted and deep, which is great in some situations but not when recording close-up vocals. Someone coughed during a recording and I was pushed back in my chair. The big bottom end plus some proximity effect, and no HPF, means some low-end filtering is going to be required at the desk. And finally, the mic has an on-axis sweet spot that might need to be explored to keep levels and proximity consistent. A large rectangular diaphragm made up of four rectangular elements cannot be expected to have the relatively even address characteristics of a circular diaphragm, and movement across the horizontal axis does not create as much change as in the vertical plane, but for best results this mic needs to be pointed at the source. GET CRACKING
The clarity on offer was perhaps more noticeable when using the AT5040 to record acoustic instruments. The mic is sometimes placed further from the source than for recording vocals but again the sound seemed drawn towards the mic. This combined with fast, detailed transients and pure, accurate mids resulted in a very natural sound. As with the vocals, you can be confident it can get a good sound from any instrument. In some ways the AT5040 gets better at greater distances. The sweet spot that is noticeable up close disappears with the mic further away from the source, while the strong and extended low frequency response keeps the sound full even when it gets a fair way from the source. And it’s almost eerily quiet. Cranking it up in the control room with the doors open and no-one in the studio I could hear my children playing outside and distant sheep (I live in the country) but no noise from the mic, just a wonderful sense of being in the space. I tried this with other mics in the room but they were all noisier and less natural.
“
It’s true that different mics suit different voices but I’m sure this would get a useable-to-great result on any vocal
Despite having to consider those details, once you start recording, the AT5040 has a beautiful tone. It’s deep and thick, breathily warm and intimate on quiet sections, kind to harsh female vocals and it doesn’t emphasise sibilance. It’s definitely flattering. And because of the way it brings the voice forward, it’s happy with the source slightly further away than other mics. The impressions I’d gained from using it live remained, but using it in the studio — where it’s quiet and you can hear everything clearly — really highlighted the mic’s abilities. Its most impressive attribute is its presence or clarity. By presence I mean the ability to pull the sound in towards the mic, to get a sense that the voice was right in front of your face, all warm and real. I liked it on every voice I heard. If it was used in the same room with other mics I could always tell which it was by the sound, and it always sounded closer than the others. It’s true that different mics suit different voices but I’m sure this would get a useable-to-great result on any vocal.
The AT5040 is sold as a premium vocal mic, and it comes at a high price compared to other AudioTechnica products, but it’s on par with equivalent high-end products from other manufacturers. Fitting a HPF and a pad are the obvious suggestions for improvement. Like all Audio-Technica mics its build quality is very high and the AT5040 has the distinction of being hand-assembled and individually inspected. They ship with a signed frequency response plot for each microphone. The AT5040 will probably live in a nice studio somewhere but the supplied hard-shell case looks road-ready should travel beckon. This mic should find a place in high-end studios wanting a modern, technologically-advanced mic to offer customers, or AV with producers who would appreciate its reach and low noise… or it could be used as the one top-quality mic in a project or voiceover studio. Innovative, and very impressive.
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By acting now you can be confident for the future and you will save money. Conditions: 1. This offer is valid for all trade in’s occurring between 1st October 2013 – 31 January 2014. 2. Access to the extra discounts applies for channel-for-channel trades of complete systems. 3. All trade in systems must be in complete working order. Non-complete or non-working units will not be accepted.
AT 40
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REVIEW
Surrounded: Reason can now control the instruments that inspired it!
PROPELLERHEAD REASON 7 After almost 13 years of careful and considered development, Reason’s ‘walled garden’ is entirely open.
NEED TO KNOW
Review: Derek Johnson
PRICE $499 CONTACT Electric Factory www.elfa.com.au sales@elfa.com.au (03) 9474 1000
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PROS Reason is now a no-holdsbarred DAW
CONS More assignable knobs on the EMI would be nice
EMI: simple, but effective.
As would more control over the ‘slice’ feature
Love the Audiomatic Retro Transformer
SUMMARY Only Propellerhead knows where it’ll go next with its flagship. As new features are added, and existing tools refined, there seems an inexorable move to matching, in its own way, the facilities of competing DAWs. But this gradual move to ‘full-on DAW’ has not been at the expense of usability — Reason still feels as musician-friendly as it ever did. May it ever be so.
In retrospect, the release of Reason 6, with its fully integrated audio tracks, heralded Propellerhead’s move to answer critics of its closed approach. Then with v6.5, last year, the wall came down, and the previously locked rack of virtual synths and effects was opened up to third-party plug-ins. There was almost nothing left for a Reason 7 update to add… except for MIDI Out functionality. And the Swedes haven’t disappointed. Finally, Reason has opened right up.
in the face of the creative software tools offered by the likes of Reason... Now look at the geekily attractive press shot of Reason linked to a studio full of that very same inspirational gear!
THE ‘MISSING’ UPDATE
EMI is as simple as it comes: it transmits data on one MIDI channel on one MIDI output, and is equipped with pitchbend, mod wheel and one assignable controller (and one program change). It’s equipped with CV controls so any device in the rack can be routed to the outside world (albeit monophonically). It can be played polyphonically (as long as the voice on the target MIDI channel is polyphonic) from a master controller assigned to a sequencer track. Complicated MIDI data, such as Bank Select commands and deep controller data, will have to be recorded or drawn on sequencer lanes. Velocity and aftertouch data are transmitted through the EMI, and recorded into the sequencer, as you’d expect.
To bring us up to date, we should take a trip past v6.5. For a normal software house, this update would have been worth a full integer, and inspired a full review. But Propellerhead isn’t like that! For the longest time, Reason was a closed system. This was fine: files could easily be swapped between users and Propellerhead didn’t have to worry about accommodating other developers’ code. I’ve always been on the fence: as much as I’d like to use my favourite virtual instruments and effects inside Reason, the ReWire interapplication protocol has usually been enough to let me link Reason to a VST-compatible host. And I’ve always enjoyed the challenge of bending Reason devices to mimic particular instruments and effects. But as of v6.5, Propellerhead’s attitude changed, with the introduction of Rack Extensions. This tightly written plug-in format is obviously designed to slot seamlessly into Reason, complete with all the established audio and CV connections, and automation control of standard devices. The SDK is free to genuine commercial developers, and some hefty names have come to a fast-growing party: Korg, FXpansion, Rob Papen, iZotope and u-he are just a handful of examples. Further control is maintained by distributing Extensions through a very 21st century integrated web store. New Rack Extensions can only be found, bought and installed from the Propellerhead Store. Freebies have been rather thin on the ground, as opposed to the vast sea of gratis VST plugs, but you can download 30-day trials if you’re collaborating with someone and missing an extension. A NEW FRONTIER
The opening up of Reason to the outside world is practically complete with the launch of v7. The big ticket item here is the new External MIDI Instrument (EMI) device: Reason can now play and control your external hardware synths. In fact, through virtual ports on your computer, it can also play soft synths hosted in other applications. This latest enhancement engenders a peculiar feeling of satisfaction. Anyone with a history with, or experience of, any generation of hardware synth will find functionality in Reason that sparks ideas that would sound even better if the other gear in the real world could join in. There is, however, an irony here. I can’t be the only one to have scaled back a hardware studio
Many of Reason’s controller devices — the RPG-8 arpeggiator and Matrix pattern sequencer, for example — are inspired by an earlier generation of electronica. Applying them to more modern instruments adds old-school ‘feel’ to a sonic universe that might not have expected it.
There’s no real downside to the EMI, though it does open up issues of latency that weren’t there in the closed Reason rack — MIDI going out, triggering the device, and audio coming back into Reason. Moving tracks around can solve any obvious problems, but it would be nice to see an automatic latency compensation tool introduced at some point. Other than that, there will be the issue of having enough audio inputs for your gear — or being tempted to buying hardware if you’re mainly soft!
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Many of Reason’s controller devices are inspired by an earlier generation of electronica. Applying them to more modern instruments adds old-school ‘feel’ to a sonic universe that might not have expected it
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IT’S NOT ALL ABOUT MIDI
Let’s not get entirely distracted by EMI, wonderful though it is, as it’s joined by plenty of other new features. First up, the factory sound bank has been expanded, mainly with a “new and powerful section of hard hitting drums loops”. See these devices? They’re Rack Extensions and they fit in the rack perfectly. Here, you can see Propellerhead’s own colourful Audiomatic Retro Transformer, inspired by one-click photo transformation tools. An appropriate photo illustrates each process, and although some (such as Psyche, Wash, PVC) require an audition to understand their effect, you’ll know what to expect from VHS, Vinyl and Tape. Vinyl even adds a 60Hz rumble! Above it is Korg’s Mono/Poly — the device that made my jaw slacken and hit my desk, as I fished out my credit card. It has the same classic sound, but with automation, patches, real polyphony, and a price of just US$49. The original is one of my personal favourite hardware synths, and having it in my Reason rack — arpeggiator and all — fulfils something of a dream.
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Another feature which you never knew you needed is the Spectrum EQ. This spectrum analyser with moving display visualises the frequency content of a selected mix channel or bus, anywhere in the program (you don’t have to be in the mixer window). Spectrum EQ’s on-screen controls offer a truly graphic experience of grabbing a curve — rather than knobs on the mixer — to tweak the EQ. Refreshing, and instinctive.
Version 7’s new Spectrum EQ is an odd beast: it’s not exactly a new processor, but provides a new way of looking at, and controlling, the mixer’s EQ. The analyser window is highly effective, and quickly becomes an essential tool when finetuning mixes.
REASON ESSENTIALS 2 Reason 7 has been released in parallel with Reason Essentials 2, which comes with Propellerhead’s Balance audio interface, and an upgrade path to the full version of Reason if required. Essentials is limited in many ways — fewer devices, reduced mixer — but offers just enough of its parent to spur creativity. Version 2 also adds audio slicing and quantizing to the sequencer, and you can create REX files from your audio recordings inside the software. The new audio import options have also been added to V2. I was surprised to see Rack Extensions supported. The cynical view might be that adding Rack Extensions generates income, but it does mean you can expand your Essentials if you don’t have any need for a full Reason 7 package.
One simple but immensely helpful tweak is support for more audio formats — including MP3, WMA and AAC — that can be directly imported into Reason. The minor hassle of converting, for example, MP3 before importing is now a thing of the past. More significantly, audio tracks (which can already be time-stretched against tempo) have gained a ‘slice’ option. This automatic process slices a track at each peak, and lets you quantize, fix, tweak and generally get creative with the timing of the track. There’s not a lot of user control over the process but therein lies its simplicity. There will be audio that the process doesn’t work well on, but largely it gets the slices right. Manually removing or adding slices to a mis-analysed track can be a pain but this hasn’t happened too often for me so far. At the moment, only one track at a time can be processed and tweaked, and the process doesn’t include any pitch tools on a per-slice basis. You can manage most of what you want using other Reason tools, though.
Beyond the automatic time-stretching on the post-v6 audio tracks, a ReCycle vibe has been added to audio tracks with a slice option that detects peaks automatically and places slices at the most obvious points — it’s nearly always right! Then quantize, re-time, or export the result to REX-compatible devices. AT 44
If this is sounding something like an in-line version of Propellerhead’s Recycle slice and dice loop tool, then you won’t be surprised to hear that sliced tracks can be converted to Recycle’s REX file format. The exported file can be used elsewhere or kept in-house for playback and treatment through the Dr:Rex or Dr Octo Rex devices, any of the Reason sample players or the Kong Drum Designer. MIXING IT UP
Reason’s mixer is already tightly integrated with the rack. But Propellerhead has taken it to another level. First of all, mixer channel level and pan controls now appear in the rack itself, so you don’t have to keep jumping to the mix window. The mixer also features new bus channels, which make it easy to group tracks for mixing big sessions. Apparently, setting up one audio track for parallel processing in a pair of mixing channels is now a ‘thing’ with a 21st century name — ‘New York compression’. Well, this technique is as easy as a right click on a mixer channel, applying whatever contrasting signal processing you’d like to experiment with on the channels and mixing the result. This is much fiddlier in other soft- and hardware situations.
And that is not all: v7’s Audiomatic Retro Transformer is a deceptively simple tool that is now always in use somewhere in my Reason work. It offers 16 ‘snapshots’ — including VHS, Vinyl, MP3 — that impose the psycho-acoustic character and frequency response on to the selected track. I love this even though I know I shouldn’t. The control here seems limited but it does all it needs to. Actually, the Audiomatic Retro Transformer is a Rack Extension; free with Reason 7, users of V6 or V6.5 can buy it for US$49. OPEN ARMS
After 13 years — an eternity in software terms — of gradual evolution, Reason has matured and opened itself up. Okay, there’s no VST support, but that is even more unlikely to happen than it ever was. The eternal fence-sitters should sort themselves out now that the rack is open to thirdparties and external MIDI gear. I wonder about surround mixing and monitoring but don’t yet miss it myself. If there’s anything you’d like to see working in a certain way, tell the Swedes. They do listen to the steady stream of constructive feedback generated by their users, and come up with considered, streamlined solutions to some of the problems raised. That’s why Reason keeps growing: because people like Propellerhead’s approach, and the open-ended way it allows users to do their own thing. And those hundreds of thousands of Reason users are happy to follow that lead. This upgrade comes highly recommended.
The External MIDI Instrument device completely opens up Reason to the rest of the world, connecting Reason to whatever sound makers you have on your keyboard stand or in your rack. Audio is routed into Reason’s mixer via the usual channels (and the hardware channels of your audio interface) and MIDI heads out of your MIDI sockets — one channel at a time, obviously. If your external device is polyphonic, then it can be played that way from your master controller via Reason’s sequencer channels. If triggered from the rack by, say, the Matrix step sequencer, life is refreshingly monophonic. Automation of the external MIDI device is possible, and program changes, CC automation, and pitchbend, mod wheel, velocity and aftertouch can all be sent out of the EMI. The device also has one assignable controller knob on its front panel.
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REVIEW
FITS IN YOUR HAND The H6 is roughly the same width as the H4n, making it easy to grip, though comparing the main bodies it’s a couple of centimetres longer. It’s a bit top heavy (mic end), but a recessed groove underneath puts your finger in the perfect cradling possie.
THROW IT OVER YOUR SHOULDER
It’s feature-heavy for a handheld device, and with potentially six mic cables spidering out of the body you may want to use the two camera strap loopholes so you can wear it. The angled screen suits the dangling orientation as well as handheld. There’s also a tripod mount on the underside for hotshoe or DSLR cage mounting with the right adaptors.
BRUSH OF CLASS
The main body is of a similar rubberised plastic as the H4n, and it feels just as tough. On top, Zoom has traded in the scuffable painted plastic face for a black brushed aluminium look, bigger buttons and indictor LEDs that flash when clipping.
HOT MENU
The H6 features a full-colour display with plenty of detail. There are meters for each input, as well as indicators for high-pass filters, compression and phantom power. You can also see the relative fader levels and pan assignments of your monitor mix. The menu structure is roughly the same as on the H4n, except for the ability to assign compression, high pass, phantom, etc, to each channel individually.
ZOOM H6
Portable Handheld Recorder When Zoom released the H4n, it woke up a whole generation to sound. The H6 is only slightly bigger, but a whole lot better.
NEED TO KNOW
Review: Mark Davie
Price $679 (H6 w/XY & MS capsules)
Pros Better noise performance
Contact Dynamic Music: (02) 9939 1299 or info@dynamicmusic.com.au
Interchangeable capsules
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-20dB Pads, and -∞ gain controls Most issues solved
Cons Handling noise still an issue
Summary The H6 has fixed most of the issues with the H4n, including the noise, and thrown in a bunch of new features along with its well-executed interchangeable capsule system. With up to six simultaneous inputs, it’s a monster in your pocket
The Zoom H4n, and the rest of the Zoom portable recorder family has grown at about the same rate as the DSLR revolution. For all its failings — a bit noisy, push buttons for gain control, no timecode, or any other serious ENG capabilities — it became, and still is, the go-to recorder for any DSLR videographer remotely serious about capturing anything more than camera audio. And why? Well, Zoom did the one thing that other portable recorder manufacturers were unwilling to do at the time, figure out how to cram XLRs into a small form factor. It may not have been the professional practitioners tool, but it sure felt closer than plugging shotguns into a 3.5mm jack. That, and a keen price/features ratio, firmly cemented its dominance. And it wasn’t just the DSLR community that adopted the Zooms, they made their way into rehearsal rooms around the globe, and into the hands of location recorders without the budget. But as the DSLR revolution stormed on — and people began taking them into territory they were never intended to go — the performance expectation from peripherals also rose. The nub of it is: if a DSLR can be used to shoot the final episode of hit TV drama House, how far can my Zoom recorder stretch? THE GAP
There’s a big gap between the job the H4n fulfils, and the professional location recorder/ ENG kit from the likes of Sound Devices or Nagra. Reliability is perhaps the biggest issue: professional connectivity is a lot more than just XLRs, multiple outputs for interruptable foldback is a must for ENG, then there’s sync and timecode. Roland, Fostex and Tascam have done an admirable job of filling the middle market, but has traditionally stuck to one of two form factors. Handheld recorders come with a couple of inputs, but anything above that and you have to carry it over your shoulder. There’s been nothing with four or more XLR inputs you can cradle in your hand, until now. Just like DSLRs have given every videographer a shot at getting a great picture, Zoom’s new H6 gives you a genuine shot at getting great sound. The scenarios where more than two XLR inputs might be required on location are many. For instance, a simple interview might require two lavalier mics and a shotgun, or three omni handhelds if you’ve got two interview subjects. Capturing sound for a short film on location? You’ll need at least a handful of inputs for that. But best of all, the H6 can double as a multiple input interface back in the studio. H4N REVISITED
In the editorial office, we’ve been using the H4n for years now and there were a few frustrating niggles that we hoped would be looked at on the H6. Firstly, the power slider. While handy, in that it requires you to hold it in position to
either turn it on or off, on the H4n the switch was glued/snapped on from the outside of the case, meaning it could fall off. The H6’s slider looks firmly located within the case and unable to be dislodged. Good start. Secondly, gain for each channel was only accessible via a single plus/minus push button control on the side — usable, but not a great level of control, and adjusting gain while recording resulted in audible clicks in the recording. The H6 has independent rotary gain controls for each channel, including one on each of the interchangeable capsules, with enough resistance to hold their position and side guards so you can’t easily bump them off settings. Another issue with the H4n was a lack of pads, combined with the gain on the XY stereo mic going from +7 to +47dB. This meant that if the mic pickup was too hot for the preamp, there was nowhere to go. On the H6, Zoom has implemented -20dB pads on each of the combo inputs, so the preamps can receive a maximum input level of +22dBU without clipping. Better yet, the gain control on each of the capsules now goes down to -∞dB. While it does make it tricky to set low gain levels with such a steep rolloff at the bottom end, it does at least give you a chance of capturing any extremely loud sounds. This also means you can send line level signals to the H6 with the pad engaged. Saying that, be aware that it doesn’t bypass the H6’s preamps, so isn’t quite the recording solution sound operators who already owned mixers were hoping for. The fourth concern was bootup time with bigger cards. Some people were reporting bootup times in excess of a minute with 32GB SD cards. The H6 has no such issues. Whacking in a random 8GB card, as soon as the H6 booted up (a couple of seconds), it was immediately ready to record. The card already had other data on it, but the H6 just plonked its file structure amongst it and was ready to go, no re-format required. Another big bummer, and one more specifically related to the DSLR crowd was the combination headphone/line level output. You had to choose whether you’d use one or the other. On the H6, these have been separated. The line level output is still on an unbalanced 3.5mm stereo minijack, but it’s designed to plug into a DSLRs audio input, which are typically the same. Handling noise is still a major issue with the H6. While it’s possible to adjust gain with the rotary dials, you have to be careful not to wreck the take by bumping the unit. Battery life has jumped up markedly. The H6 can go for almost 10 hours with all six inputs recording, whereas the H4n could only manage about six with less inputs. And while the H4n seemed to frequently achieve less than that mark, the H6 was still on two out of three bars after using the recorder on and off for a couple of weeks.
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The shotgun and MS capsules, and the dual input add-on round out the H6’s arsenal.
SENNHEISER’S WIRELESS MASTERPIECE
SWAP ’N’ GO
The most noticeable addition to the portable recorder is the option to swap out microphone capsules. The standard near-coincident XY pair is reprised with bigger capsules, and can be swapped out for a mid-side configuration or a shotgun. Rounding out the lineup is the option to clip on two extra combo jack/ XLR inputs, taking the input count to six. It’s these choices that demonstrate Zoom’s dedication to spreading its wings as far as the user will take it. A mid-side configuration is great for a singer-songwriter to capture a guitar in stereo with a separate vocal mic plugged in. And a shotgun is perfect for the DSLR market. Or if you need more inputs just grab the extra input module. (The H6 comes with the XY and mid-side capsules, while the shotgun and input module optional extras.)
DIGITAL 9000
The capsules clip into the body securely with side pushbuttons to release them. It all feels very solid except for the plastic connecting the MS capsule ballgrille housing to its base. It feels like it would be the first thing to go in this configuration, but I wasn’t game to test its rigidity. When the mid-side capsule is plugged in, apart from being able to adjust the overall input level with the main gain knob, the menu navigation toggle automatically senses the different capsule and becomes a secondary adjustment over the amount of side information captured. You can go from ‘off ’, which gives a simple cardioid pattern, through -24dB, up to +6dB for ultra-wide, or just switch it to a raw capture into the left and right streams (cardioid on left, bi-directional on right) to be decoded later. As you toggle through the values, the size of the figure-eight overlay changes to suit, which is a nice touch. NO NOISE IS GOOD NOISE
Functionality aside, the main hope for the H6 was that it would improve on the preamps of the H4n. The H4n specs never quoted a noise figure, but it’s appreciably high. The H6 has greatly improved on this, quoting an equivalent input noise of -120dBu or less on each of the XLR inputs. It is significantly less noisy at higher gain levels, especially noticeable when using shotguns or low level handhelds.
Fully Digital Rock Solid RF Pure Audio
The XY mic capsules on the H6 are noticeably larger than the H4n’s, with 14.6mm diaphragms. It’s still well in ‘small diaphragm’ territory, though, and there’s not a major difference between the two. If you had to split it, the H4n was a hair’s breadth more responsive to transients, the H6 a little more subtle.
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The mid-side capsule works a treat, and the extra flexibility of adjusting the sides or sticking with a mono capture in post is handy. Unfortunately, the shotgun capsule wasn’t shipping at the time of the review, so we couldn’t test it. It’s an interesting design, though, WWW using three capsules and digital signal processing to yield a hypercardioid pattern instead of the standard interference tube design.
AUDIO SAMPLES Listen to a comparison of the preamp noise levels on the Zoom H6 and the H4n. Voce recorded with a Shure SM63 omni dynamic handheld.
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CLASS H
The H6 is a dramatic improvement on the H4n. It’s obviously not meant as a strict replacement, as Zoom continues to sell the H4n and it serves a big slice of the market. That said, Zoom has managed to give a dedicated following more of what it wants, and it’s hard not to see people upgrading. The interchangeable capsules are a real value boost in an already packed unit — well worth it if you need more inputs and flexibility, especially without sacrificing much size.
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REVIEW
ZYNAPTIQ UNFILTER & UNVEIL PLUG-INS Unfilter and Unveil are the Autotunes of excessive filtering and reverb, and that’s a good thing. Review: Dax Liniere
We all baulk at the idea of ‘fixing it in the mix’ but sometimes that’s exactly what needs to happen. As mix engineers, occasionally we receive tracks that aren’t quite up to scratch. Perhaps the vocals are noisy or have comb filtering, the drum overheads sound like room mics, or somebody’s applied a weird filter effect to a drum loop.
filtering — all in real-time. The incoming signal is analysed and re-equalised based on the detected frequency response. Highly resonant or heavily band-passed material is effortlessly repaired with the turn of a single knob. Examples on the web were startlingly good, so naturally, I had to try it out myself. I recently mixed a heavy metal track that had been recorded in another studio. The main vocals were heavily processed with a band-pass filter (ie. a telephone effect). I felt this suited only a few parts of the song. Inserting the plug-in, I simply ‘unfiltered’ the audio. It didn’t become Frank Sinatra, but I could now achieve exactly what I wanted. Sure, you might achieve a similar result with regular EQ, but it would take a lot of time and I’m willing to bet the result would be inferior.
Sampling noise reduction, spectral repair and Autotune are some of the tools that allow us to deliver an acceptable final product when the odds are stacked against us. But it’s only every so often we get new, innovative and powerful tools that change what we can achieve with the material we’re given. Unfilter and Unveil, the latest offerings from plug-in developer Zynaptiq, fall right in that game-changing category. The name may sound new, but Zynaptiq’s Chief Technical Officer, Stephan Bernsee, has been developing DSP application since before he founded Prosoniq in 1990.
Tip: For non-wideband sources (vocals, individual instruments, etc) you will need to tailor the desired frequency response with the built-in EQ or Intensity Bias modes.
Unfilter is an intelligent EQ capable of correcting frequency response anomalies such as resonances, cut-off filtering, extreme EQing, and even comb
Unveil is designed to control reverberation and signal focus (similar to image blur removal) using a proprietary artificial intelligence-based technique.
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Its main purpose is removing reverb and ‘mud’ from any material, including music and dialogue — and it does that very well. Excess room sound in source tracks is a problem all mix engineers have faced at some time and previously, there was little you could do. Transient designers and gates work with limited success and the end result is usually a compromise. Along comes Unveil and, once again, we have a gamechanger. Distant-sounding drum overheads can rob a performance of its immediacy and punch, but thankfully, Unveil can dial it right back in. It’s not going to dry up the Grand Canyon of reverbs, but most of the time you just need a few dB less reverb to save your bacon. Extreme settings can result in unnatural, ‘pokey’ transients. However, just as Autotune was originally intended as a corrective tool, Unveil can be used creatively, too. If you mix for a living and want to impress your clients, these tools will certainly give you a competitive advantage. US$399 each at www.zynaptiq.com
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