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Meet the SoundField By RØDE DE NT-SF1: NT T-SF1: the most versatile microphone ne in the t world.
ONE MICROPHONE. ALL MICROPHONES. The SoundField by RØDE NT-SF1 is possibly the only mic you u will ever need. Coupled with the SoundField by RØDE Plugin, it can become a urround feed, mono shotgun mic, a stereo g-8 main pair, a complete 5.1 surround a hyper-realistic 7.1.4 with-height audio capture device, or a head-tracked 360° soundscape recorder for virtual reality. Whether you are recording music, broadcasting, capturing location sound or working at the frontiers of w, unlocking 360° video production, the NT-SF1 will enhance your workow, unlimited creative possibilities.
SoundField by RØDE Plugin Experiment with different mic orientations, polar patterns and congurations in post-production.
Based on an all-new, ultra-low noise true-condenser capsule and with RØDE’s legendary sound quality, the NT-SF1 comes with our 10-year industry-leading warranty and all accessories, including cables, shockmount, blimp and furry windshield.
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Editor Mark Davie mark@audiotechnology.com.au Publisher Philip Spencer philip@alchemedia.com.au Editorial Director Christopher Holder chris@audiotechnology.com.au Assistant Editor Preshan John preshan@alchemedia.com.au Art Direction Dominic Carey dominic@alchemedia.com.au Graphic Designer Daniel Howard daniel@alchemedia.com.au Advertising Philip Spencer philip@alchemedia.com.au Accounts Jaedd Asthana jaedd@alchemedia.com.au Subscriptions Sophie Spencer subscriptions@alchemedia.com.au
Regular Contributors Martin Walker Paul Tingen Guy Harrison Greg Walker Greg Simmons Blair Joscelyne Mark Woods Chris Braun Robert Clark Andrew Bencina Ewan McDonald Cover Photo Daniel Linnet
AudioTechnology magazine (ISSN 1440-2432) is published by Alchemedia Publishing Pty Ltd (ABN 34 074 431 628). Contact +61 3 5331 4949 info@alchemedia.com.au www.audiotechnology.com.au PO Box 295, Ballarat VIC 3353, Australia.
All material in this magazine is copyright Š 2018 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. 24/10/2018.
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COVER STORY
Studios 301 is Back, At What Cost?
16
ISSUE 51 CONTENTS
22
Mix Masters: Björk Empowers Marta Salogni
Quick Mix: Callum Rendell
14
Avid Pro Tools | MTRX Audio Interface
Universal Audio Arrow Interface AT 6
52
View from The Bench: Tracing the EQ Path
30
Sony’s Hi-Res Mics Go to 50kHz
42
PMC Result6 Shortens Path to Transmission Line
28 48
46
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GENERAL NEWS
MOOG ONE Moog One is the company’s first polyphonic synth in over 35 years. The tri-timbral analogue dream-synth is what Moog calls its most ambitious project to date — an electronic odyssey encompassing years of research and decades of exploration, culminating in the birth of a new flagship synthesizer. Available in eight-voice or 16-voice configurations, each individual Moog One voice circuit contains an analogue signal path more powerful than a Minimoog Voyager, harnessing the power of three newly-designed dual-output analogue VCOs with ring mod and FM, two independent analogue filters, a dual source
analogue noise generator, analogue mixer with external audio input, four LFOs, and three envelope generators. Each of Moog One’s three analogue timbres is an independently-addressable polysynth, each with its own sequencer, arpeggiator, and onboard effects library — including a suite of professional reverbs from Eventide. Innovative Music: (03) 9540 0658 or info@innovativemusic.com.au
SSL FUSION Fusion is a new all-analogue 2U outboard processor from SSL that’s designed to add tonal character, weight and space to your mix bus or stereo stems. Fusion is comprised of five analogue processors. Vintage Drive is a unique non-linear harmonic enhancement circuit with Drive and Density controls to produce harmonic saturation and soft compression. The SSL Violet EQ is a minimum phase-shift, two-band shelving EQ which draws on the SSL legacy of carefully selected frequencies and response curves to create musical and intuitive tonal control. A new High Frequency Compressor affects only the highs and is optimised
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for smooth and transparent harshness reduction and a tapelike high-frequency roll off. A Stereo Image enhancer provides a true analogue mid-side circuit that allows for widening and spatial manipulation of the stereo field. Finally a switched SSL Transformer circuit introduces subtle low-frequency saturation, alongside a high-frequency phase-shift. Amber Technology: 1800 251 367 or sales@ambertech.com.au
MORE NEWS AT www.audiotechnology.com.au
API 1608-II CONSOLE API’s second version of its 1608-II console stays true to API’s heritage of crafting desirable analogue consoles. Rather, API’s engineering team focused its efforts on making the 1608-II even more intuitive to use, including a layout upgrade to the auxiliary masters, the inclusion of four automatable stereo returns, and a revamp of the control room section of the console. “Also new to the 1608-II is API’s optional cross-platform Final Touch automation system,” says API Director of Engineering, Todd Humora. “Final Touch is a step up from the already excellent automation system that was offered with the original 1608.
Moving forward, Final Touch mixes can also be shared among API consoles that feature Final Touch, such as Legacy AXS or Vision consoles.” API also released the 2448 console to plug the gap between the 1608-II and AXS consoles. It has 24 fully featured channels for recording, and 56 channels for mixing. It also has four stereo returns and can be optioned up with Final Touch. Studio Connections: (03) 9416 8097 or enquiries@studioconnections.com.au
RODE NT-SF1 360° MIC The Røde NT-SF1 is finally here — the company’s first broadcastgrade ambisonic microphone since it purchased SoundField. What’s ambisonics? The gist is, it’s using an array of microphones — a tetrahedral array, in the case of the NT-SF1 — to record in a spherical pattern around the microphones. You can then decode the signal of all those mics, and create any polar pattern you like. It’s very useful for capturing environments in film and indispensable if you’re engaging with VR content. The huge news about the NT-SF1 is Røde has managed to bring the price down under US$1000; an unheard of price point for SoundField
microphones. Not to mention it comes with a blimp, the proper 4-into-1 cable you’ll need, and a custom suspension mount. Decoding is handled by a free plug-in called ‘SoundField by Røde’. It’s perfectly matched to the new mic and decodes the ‘raw’ A-Format into a more workable B-Format. Rode: (02) 9648 5855 or info@rode.com
MORE NEWS AT www.audiotechnology.com.au
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LIVE NEWS
PRESONUS CDL POINT SOURCE ARRAY The team at Presonus and Worx Audio have released two new active loudspeakers — the CDL12 hybrid point source array, and CDL18S subwoofer. Presonus says the CDL12 combines directivity of a point-source design with the scalable coverage of a line array to create a unique loudspeaker that provides the flexibility demanded by modern sound reproduction applications. The Dante-ready CDL12 incorporates eight twoinch high frequency drivers positioned array-style down the middle, in front of a single 12-inch woofer with a 2.5-inch voice coil. The Class D amp provides 500W RMS of power with a
crossover at 420Hz. Nominal coverage is 120° x 20°. Onboard DSP presets cover compression, limiting, delay and eight-band EQ. The CDL18S subwoofer has a single 18-inch LF transducer with a four-inch voice coil. Class D amplification provides 500W RMS. The sub is also Dante enabled and the integrated pole mount lets you sit up to two CDL12 boxes on top of it for use as a portable system. Link Audio: (03) 8373 4817 or info@linkaudio.com.au
D&B GROWS SL-SERIES Not long ago, d&b audiotechnik established its SL-Series with the flagship GSL line array system designed as a complete package for arena, stadium and festival environments. Engineered to deliver precise broadband directivity control across the entire audio bandwidth, GSL’s hallmarks are quiet stages for the performers, maximum art for the audience and minimal noise for those beyond. Now d&b adds to the SL-Series with a smaller system sharing the same DNA — KSL. Due for official release on 24th January 2019 at the NAMM show, KSL delivers all the recognisable features of its SL lineage from broadband directivity
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to advanced rigging options and can be used as a standalone, self-contained package, a delay, or fill accompaniment to GSL, maintaining the SL-Series family attributes from top to toe. The introduction of KSL expands out the SL-Series’ feature set into wider areas and events, including performance installations of any genre. NAS Solutions: (03) 8756 2600 or sales@nas.solutions
MORE NEWS AT www.audiotechnology.com.au
QSC CP SERIES QSC’s new CP Series of portable powered loudspeakers is comprised of a pair of two-way models, the CP8 and CP12. Both models feature a highly efficient 1000W Class D power module, one-touch preset DSP contours for common applications, and line, mic/line and 3.5mm stereo inputs. Each can also be polemounted, utilised as a floor monitor, or deployed in a fixed or temporary installation. Available accessories include carrying tote, outdoor cover and quick-connect yoke mount. “QSC is known the world-over for delivering quality, high performance products for professional use,” states QSC David Fuller, Sr.
Director, Product Development. “The ability to deliver that same value proposition to a far broader range of customers is a remarkable achievement for our teams.” Fuller adds, “Likewise, CP offers an incredible introduction to the QSC brand for first-time buyers, as well as a long sought-after QSC solution for commercial customers.” Jands: (02) 9582 0909 or www.jands.com.au
ADAMSON GOES TO MONITOR CITY Melbourne production company Monitor City has made a major investment in a new Adamson audio system sourced through CMI Music & Audio. The system is built around Adamson’s flagship E-Series and comprises a complement of E12 three-way, full-range line array enclosures and E119 subwoofers. Rounding out the package are the compact S10 two-way, full range line array enclosures. Adamson’s touring E-Racks drive the system, each equipped with three Lab.gruppen PLM+ series amplifiers featuring Lake processing and Dante audio networking functionality. Monitor City Co-Founders Adrian Barnard and
Matt Dufty carefully evaluated all the major players in the PA market before deciding on Adamson for their purchase. “We looked at factors such as the speed of deployment, the logic of the system, its capabilities, and sonic performance,” says Dufty. “Adamson stood out as the best choice. This system will cover our festival work in the upcoming summer months, and can then be split into multiple systems.” CMI Music & Audio: (03) 9315 2244 or www.cmi.com.au
MORE NEWS AT www.audiotechnology.com.au
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SOFTWARE NEWS
BIG NI UPDATES Native Instruments Komplete range of software has been updated to version 12, and it’s a doozie.There are four versions: Select, Standard, Ultimate, and an all-new Ultimate Collector’s Edition that includes the full Symphony Series collection. Inside, Kontakt has been updated to version six with a wavetable engine, new instruments include TRK-01, sounds of the Middle East, and Thrill. You also get access to Massive X when it come out in 2019. The highly anticipated Komplete Kontrol S88 Mk2 controller has a brand new Pro-grade Fatar fully-weighted keybed and comes bundled with Komplete Select and Maschine Essentials software.
Expect to pay $1399 for an S88 Mk 2. If you find S-Series keyboards a tad pricey, NI has put out the A-Series range. These USB2 bus-powered keyboards come in 25-, 49-, and 61-key sizes and start from a street price of just $229. Finally, Maschine Mikro Mk 3 is a cut down version of the full-scale Maschine controller sporting 16 RGB-backlit pads, a dual-touch Smart Strip and various beatmaking controls. It will go for a street price of $379. CMI Music & Audio: (03) 9315 2244 or www.cmi.com.au
SSL FLEXES NEW LOOK REVERB Solid State Logic’s Native plug-ins have been undergoing a design overhaul to give the virtual faceplate and knob design a decidedly more SSL-ish look. The latest candidate is its reverb. FlexVerb is a free cross-grade for pre-existing X-Verb users, and part of the new v6.1 software release. If you already have SSL Native v6, you get this for nix. FlexVerb features deep control over your reverb path. You can choose to split your early reflections and reverb tails for some interesting hybrid effects; say, a plate reflection into a large hall tail. There’s also a six-band EQ onboard, to sculpt your output. On top of that, there are three bands of reverb time multipliers, so you can really get the tails swimming in one part AT 12
of the frequency spectrum, but dial it back in others. Lastly, there’s also an input side chain to make it dead easy to get the reverb out of the way of your source. Version 6.1 also improves the preset management system, and adds dedicated MCU control mapping for Logic and Cubase. If you’re not already an SSL Native user, you have the choice of subscribing for a monthly fee of US$14.99 or buying the plug-ins outright. The complete bundle is US$1599. Amber Technology: 1800 251 367 or info@ambertech.com.au
MORE NEWS AT www.audiotechnology.com.au
TC ELECTRONIC DVR250-DT The TC Electronic DVR250-DT is part of TC’s Icon series of hardware-controlled effect plug-ins and brings the classic reverb sound of the EMT 250 Electronic Reverberator to your DAW. Like the original, the DVR250-DT marries hardware control with software processing and sounds great on anything from lead vocals and synths to drum hits and percussion. TC says the DVR250-DT is the perfect match for any track “that needs sonic perfection and a warm vintage vibe.” The hardware-controlled
plug-in is available in VST, AU and AAX formats and includes built-in echo, delay, chorus, phasing and space effects. TC also released the new TC1210-DT spatial expander and stereo chorus flanger, as well as the TC8210-DT reverb, both of which are modules for its hardware-controlled plug-in range. Amber Technology: 1800 251 367 or sales@ambertech.com.au
WAVES ABBEY ROAD TG PLUG-IN The distinct solid-state transistor-based sound of the EMI TG12410 Transfer Console at Abbey Road Studios has proven itself time and again on albums like Pink Floyd’s The Dark Side of the Moon, Nirvana’s In Utero and Radiohead’s OK Computer. Developed in conjunction with Abbey Road Studios, the Waves TG Mastering Chain plug-in bring the same magic to your DAW. Just like the original console, the TG Mastering Chain plug-in is made out of modules (or cassettes): the TG12411 Input Module, TG12412 Tone Module (EQ), TG12413 Compressor/Limiter Module, TG12414 Filter Module, and TG12416 V.A.L (Spread)
Module (stereo component only) incorporated into the Output Module. Modules can be interchanged and switched on/off to allow for a flexible processing flow and creating custom chains. Individual modules can be used on different tracks and group buses when mixing, or on the master bus for mastering. Sound & Music: (03) 9555 8081 or www.sound-music.com
MORE NEWS AT www.audiotechnology.com.au
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QUICK MIX
The
with
Callum Rendell
Who are you currently touring with? I am currently on the road with Brisbane band, The Brave. What are some other acts you’ve worked with? My permanent act is Melbourne band The Beautiful Monument, but I’ve worked with my fair share of local and international touring acts, recently with the likes of Being As An Ocean, Slaves, Polaris and Knocked Loose. What was your path to a career in audio engineering? I’ve been mixing bands around the country for around two and a half years now. I started off in my hometown rolling leads for the local production company at NYE events, before moving to Melbourne to study audio at RMIT. What is your favourite console and why? Digico’s SD series have always been personal favourites of mine. Once you figure out the workflow, they are very fast and easy to use, plus they sound incredible! Favourite microphone or any other piece of kit? Shure Beta91A is probably my most used piece of gear. I mix heavy/ rock bands the majority of the time, so it’s a pretty crucial part of my kit! Most memorable gig or career highlight? Probably mixing FOH and five sends of IEM for The Beautiful Monument at Unify Festival this year. Mixing on such a big (and beautiful sounding) PA to such a massive crowd was an experience! What are three mixing techniques you regularly employ? Compression, parallel compression, and group/bus control. What are three pieces of gear or features that have been game changers for you? Waves integration for smaller consoles has been huge. Running a full complement of top end plug-ins on basic consoles changed the way I mix. Being able to scan new channels on Sennheiser G3 IEM units quickly has also been a massive time saver when we walk into a venue or festival. Tablet connectivity also plays a massive role in most of my work, being able to get out of the corner of the room and mix in the crowd is huge!
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How have your working methods changed since you began live sound mixing? It’s hard to compare mixing on an eight-channel analogue console in the back of a pub to the SD10 at The Corner Hotel. The use of groups and busses made a massive difference, especially trying to achieve the polished and ‘larger than life’ drum sound heavy bands are searching for. Any tips/words of wisdom for someone starting out? Get out there and take as many gigs as you can get and meet as many people as possible. You’ll always learn more on the job than at a school, and the people you meet in the real world can be crucial to what you do later on. It’s all about who you know!
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FEATURE
STUDIOS 301 IS BACK! Tom Misner has rescued Studios 301 from extinction and kept its run as Australia’s longest-operating studio going. Will it be more than just a rich man’s hobby project? Story: Mark Davie
Some blokes have luxury cars, some blokes have real estate, some blokes have studios, some blokes have the lot. That’s Tom Misner; SAE founder, Studios 301 owner, and resident of the millionaire state, Monte Carlo. “I have all the luxury items in the world you can think of, but I still like building studios,” mused Misner. “It’s weird.” It’s only weird because it’s a hobby the average income earner can’t afford. It’s his version of a model train set; impeccably sculpted to his specifications, with analogue consoles and vintage mics replacing period-correct station facades and heritage-variety plastic shrubbery. Misner bills the new incarnation of Sydney’s Studios 301 as a philanthropic gesture. “I started making money in Australia, and when I sold SAE I made even more money,” he said. “This is like me giving back something.” He’s right in that there’s little expectation Misner will bank the sort of return a $12 million investment would typically net him. In his view, it’s philanthropic because building a world-class studio in Australia that no one else could afford to fund, will be a leg-up for local producers and engineers; allowing them to work at a “higher calibre”. In his regional Monaco parlance, he likened Australia’s lack of “truly world-class” facilities to “bringing a sports car to a Formula One race; you’re not going to win. This allows you to play on the big stage.” While Misner concedes there have been many “okay” studios in Australia, (previous incarnations of 301 included) and still are, he’ll stomp all over his past efforts on his way to telling you how much better the new build is. “The old mixing room at 301 in Castlereagh St AT 16
had an eight-foot ceiling. It was a joke,” he said. “Acoustically speaking, there’s nothing you can do with an eight-foot room. You can produce things out of it, but you have to get used to it.” Likewise there were “problems at 301 Alexandria. Things were good, other things weren’t so good. It had its run, it had its time.” At least his claims are simpler to quantify this time. Unlike decoding his ownership stake of 301 over the years or estimating his net worth before the sale of SAE, there’s literal concrete proof of the size, scale and sound of Studios 301. It is worldclass, at least on the inside. This isn’t set in the countryside like a Real World, occupying an old cathedral like Lyndhurst Hall, or stationed in a penthouse like Jungle City in New York. It’s still got to make a dollar in Australia’s toughest real estate market, which means 301 now sits at the end of a warehouse cul-de-sac, five minutes from Sydney’s International airport. What it is, is a studio with impeccable acoustics and a gobsmacking selection of gear that can handle any job, not just the big, small or esoteric ones. IN THE LONG RUN
This is version four of Australia’s longest running studio, and one of the oldest in the world. The studio began when the smaller of two gramophone manufacturers who’d set up distribution in Australia took a gambit on the local entertainment industry. The Columbia Graphophone Company thought it’d be a savvy idea to establish a studio to record local content, and set up a space on the corner of Parramatta Road and Columbia Lane in Homebush in 1926. In 1954, the studios relocated to 301 Castlereagh Street, in Sydney, where it was given the EMI name by its now-merged parent company. Eventually the studio’s management bought it out and renamed it Studios 301. In 1998, Misner bought it. 301 was then relocated to 18 Mitchell Road in Alexandria, but kept the name. The ownership gets a bit uncertain around the time Misner sold SAE to Navitas in 2011. He eventually bought it back in 2017, just as the landlords were filing DAs with the council for an apartment complex on the site, hence the relocation to the current spot at 3 Ellis Avenue in Alexandria. Misner reckons he’s built over 200 studios so far, which is probably not an exaggeration. While he may not have been as intimately involved with the others as with the current 301, he did preside over a rapidly-expanding SAE network of campuses, each with their own collection of studios. It’s his hobby; one he’s been trying to perfect for decades. He took a special interest in this build, eager to get his sweet tooth stuck into building a candy factory. As soon as the lease was signed, he immediately began toiling over floor plans. Once he’d sketched out a rough layout, the plans would circulate between himself, the architect and acoustician Jochen Veith, “but as far as details, down to the colour or choice of light switch, everything relied on me,” said Misner. “There’s nothing that can go on without me, I insist on knowing about it.” As for the colour, well, he’s gone with touches of a light Royal Blue.
Despite having the last say on every colourway and fixture in the building, this isn’t Misner’s oversized personal studio. While a big studio might be a hard beast to maintain, he’s giving it a fair run at being a commercial success. There are multiple studios (more than 301 has ever had) on a sliding cost scale; 301’s first surround post-production facility; a greater variety of production spaces; and the recording/mixing studios are just across the hallway from the mastering suites, hopefully spreading expertise.
BY THE NUMBERS $12 million 39,000 core-filled concrete blocks. 3500 sheets of marine ply 220 specialty absorbers built onsite 130 microphones 68 truck loads of concrete 42km of electrical cable
ACOUSTIC MATCHED PAIR
15km of data cable
A studio can have all the appropriate spaces, and swathes of flash analogue gear, but it’s all worthless if the rooms sound trash. Misner tapped JV Acoustics’ Jochen Veith to deliver on the promise of a world-class acoustic environment. Veith’s client list includes super producers like Max Martin and Dr Luke, Florian Schneider from Kraftwerk, Wisseloord Studios and Florian Opahle from Jethro Tull. Veith has known Misner for about 30 years, having worked together on a handful of studio builds. “He came to me and asked… actually he didn’t ask, he said, ‘We are doing a studio together in Sydney!’” From that point on, Jochen thought the balance between Misner’s design work and his mathematical approach to the acoustics was relatively straightforward and symbiotic: “We matched well actually, and it was very creative, which is the good part.” The first point of call for any studio build is checking if the building is suitable, said Jochen. Airport, railways and heavy trucks are your main concerns. Being close to an airport and in an industrial zone, it seemed as if they’d struck out on two of those. However, Veith said although the airport was close by, it’s vibrations you’re worried about. Neither overhead planes, nor the slowmoving trucks next door caused much concern. Having trucks nearby actually helped. The slab was built to handle fully-loaded semis, so loading the concrete foundation with massive concrete bunkers wasn’t a problem. “We decided to build the walls massive, in concrete block stone not lightweight plasterboard,” explained Veith. “The whole structure is decoupled from the floor which gave us the possibility to put all the airconditioning mechanicals on top of these rooms. It’s more or less a little village of upside down concrete boxes floating on the main slab, with floating floors inside the shells that are tuned to a very low frequency.” Before the bunkers get set into stone, Veith usually interviews the studio owner to figure what goes where. This time, it was a back and forth between Misner drawing out floor plan options and Veith simulating the modal fields to determine the best possible dimensions. “The whole frequency range up to 200Hz is dominated by this model field. If you get the dimensions and ratios wrong, you can’t fix it later on with acoustic room treatment. You have to be spot-on with the position of the speakers, with the listening position and with these dimensions.”
Six consoles
Acoustician Jochen Veith in the sweet spot of the Studio 1 control room.
Veith knows his limits. He’s not an acoustic shaman who can wave a measurement microphone around a room and calculate the exact Helmholtz resonator to soothe all modal inflammation. “A lot of people find it difficult to understand you can’t heal it later. Most of the time it’s nearly impossible to fix a big dip.” Normally Veith would only be able to adjust one dimension of the room; width, depth or height. Two of the three are typically fixed to a large degree. “Here, it was different,” said Veith. “It was a huge space and we could play around with it and try to find the best ratios for all of the rooms.” It was an uncommon degree of freedom, which he said could be too much at times. There was no ‘best case’ dictated by constraints; he had an open slate to build the best rooms possible. MONITOR PROGRESS
All of those dimensions mean little if not developed in concert with the monitoring solution. Deciding which speakers and whether to soffit-mount or have them freestanding helps to design the room. “It’s part of the system,” explained Veith. “It’s more like the speaker and the room is one big piece. It’s very difficult if you’re trying to find out if it’s the speaker which has a problem or the room.” What he means is, if you aren’t intimately familiar with the speakers you’re designing a room for, it can be hard to determine whether it’s the speaker that’s the issue, or the room. In the case of Studios AT 17
STUDIOS 301 FLOORPLAN The studio has a central corridor that loops in a horseshoe from the left side of the foyer, with short airlock corridors to the three main recording studios. The digital side of Studios 301 has been overhauled, with a complete backbone of Apogee Symphony MkII converters in studios 1, 2 and 3, with the latest Pro Tools 2018 installs, UAD Ultimate plug-in bundles, Waves Mercury, Fabfilter and more. They also have the new iMac Pros tethered to the Symphony units in the airlock via Thunderbolt extenders. You can simply plug in your session and start working. If tape is more your speed, you can roll in the Studer A800Mk11 24-track two-inch machine and get up and running with a simple DB25 patch. Both Studios 1 and 2 also have an Ampex ATR-102 two-track, 1/2-inch machine for mixing down to. 301 has multiple EMT plates that can be hard patched into any studio from the museum-like patching rooms. It’s also possible to route any live room patch panel into any control room, giving access to any space if it’s available.
Studio 9 (45m²)
Live Room 1 (202m²)
Studio 2 (45m²)
Studio 4 (16m²)
STUDIO 1 The centrepiece of Studio 1’s spacious 75sqm control room is the 72-channel Neve 88R console, which the orchestral crowd loves. To keep pop/rock producers happy there’s also a more complete collection of outboard that includes a Fairchild 670, multiple Urei compressors, Pultec EQs, LA2As… all the classics.
Control Room 1 (75m²)
Reception Desk
LIVE ROOM The live room attached to Studio 1 is a formidable 201sqm capable of handling a 40-50 piece orchestra and two large isolation booths attached, one specifically designed to roll the Yamaha C7 grand piano into for separation. Though it’s slightly smaller, and the ceiling isn’t as ornate as the room at Mitchell Road, its rectangular shape will yield more usable floor space and the parquetry floor is gorgeous. Getting the balance of reverb time right is dictated by the use of the room, said Veith. “There’s a big difference if you want to record pop music in smaller rooms versus recording an orchestra. In a studio, you might have a lot of different rooms with special sounds, or one or two very flexible rooms.” There’s more than one live room at Studios 301, but only one big hall connected to Studio 1, so it had to be flexible. “We tried not to be too lively, because it’s not only for orchestral music,” explained Veith. “You’re also able to record a single instrumentalist in there. As well as going for the right general reverb time, we also reduced the reverb time a little bit in the mid-frequency range around 350Hz. That way we have nice-sounding clear high frequencies, and warmth in the low end but try to reduce the cheapsounding mid-frequency range.”
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STUDIO 2 The 45sqm control room of Studio 2 houses a 48-channel SSL K Series console, and the 35sqm live space is divided into three booths. There’s plenty of classic gear installed into the racks, along with eight channels of Neve 1084 strips for those who want to augment the SSL’s front end. It’s obviously a great place to mix, and is big enough to record bands with good separation.
Live 4 (9m²)
StudioStudio 5 5 (20m²)(20m²) StudioStudio 10 10 (45m²)(45m²)
StudioStudio 11 11 (45m²)(45m²) Patching Patching Room Room
MASTERING STUDIOS The back wall is lined with three mastering studios; one for Steve Smart, one for Leon Zervos and the other shared between Andrew Edgson, Ben Feggans and Harvey O’Sullivan, each with their own complement of outboard.
Live Space 2 Live Space 2 (35m²)(35m²)
Live 4Live 6 Live 6StudioStudio 6 6 (9m²)(9m²) (9m²) (17m²) (17m²)
Live Space 3 Live Space 3 (25m²)(25m²)
3 StudioStudio 3 (45m²)(45m²)
8Live 8 Live 8 8 7 StudioStudio StudioStudio 7 (17m²) (17m²) (9m²) (9m²) (17m²) (17m²)
STUDIO 3
/ Museum CoffeeCoffee Shop /Shop Museum (400m²) (400m²)
The surround post-production suite is a new market for 301. It features monitoring controlled by an Avid S6L control surface and Avid MTRX system, with a 65-inch screen for video. The control room is similarly-sized and proportioned to Studio 2, giving consistency to mixes if engineers go between rooms. There’s also a 24sqm live room off to the side for voice-over, ADR and foley work. Alternatively, it’s perfect for the modern recording engineer looking for a state-of-the-art control surface for their in-the-box mix, with a sizeable live room attached.
STUDIO 4 Studio 4 features a Neve Custom Series 75 console in a cosy 18sqm control room, with an adjoining 9sqm booth. Perfect for production and smaller tracking sessions.
STUDIO 5 A work in progress, Studio 5 has been designated as a writing room and is already building up a sizeable collection of instruments.
PRODUCTION STUDIOS 6, 7 & 8 Adjacent to the foyer is a collection of blank, fully-treated production suites that will be available for long-term hire.
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The Studios 301 foyer is its own museum; stacked with gold and platinum records, and historic vintage gear.
301, the decision was made early on to go with a full complement of UK-made PMC monitoring. Veith has a long-standing relationship with PMC, which has as much to do with the sound of the speakers as his familiarity with how their advanced transmission line delivers low end into the room. “I can easily work with how the room reacts to them. All the acoustic treatments work well in combination with transmission line to keep it tight. There can be problems with badly-built vented box systems, so I’m happy to work with these systems.” Every room has at least one pair of PMC speakers. A pair of flagship QB1-A speakers are soffit-mounted into the Studio 1 control room front wall. Delivering on Veith’s promise that the speaker and room should work as one, sound simply emanates from in front of you when you’re standing at the console, with no perception of boxes or woofers. The stereo image is precise; the sweet spot wide enough to comfortably work on the 72-channel console; the drivers efficiently work the large space without noticeable distortion at any level; and the frequency response is incredibly consistent all the way to the back of the room. This last feature was important, because the large control room is designed not only to accomodate producers and engineers, but allow band members to set up keyboard and pedal rigs in the large rear area for tracking. In the mastering rooms, most of the Duntechs have been replaced with free-standing full-range MB3 XBD-A systems. The new surround postproduction suite has an MB3 XBD-A system in a combination of front-mounted soffits and free-standing surrounds, and there’s a smattering of twotwo.6 near fields in the other mixing and production facilities, including Studio 4 with the Custom Series 75 console. LEAVING ROOM TO MOVE
While getting the room as theoretically sound from the get-go is important, Veith knows there’s always some deviation from the models, whether it’s due to materials, construction, or some other human factor. To compensate, he always designs in the ability to tune the room when he arrives onsite. In Studio 1, while the front wall is completely solid, the rear wall is heavy on treatment. He designed tuneable Helmholtz resonators into the corners along with broadband low frequency absorption. AT 20
In the rear wall are ‘Helmfusers’, which are high frequency diffusors with holes in-between the sheets that feed cavity resonators tuned between 50 and 100Hz. He also uses heavy foil resonators to really control the low end of the room, and other low frequency absorbers above the windows on both sides of the room. “If you have a room made out of plasterboard walls, it helps a little bit with low frequency absorption, but here it’s a very solid wall,” said Veith. “On the other hand, with a solid wall you can control it very well inside with the acoustic elements. “The tuneable Helmholtz resonator is like a Formula One addition. We only use those in rooms at the top class, only in the frequency range between 25 and 50Hz, and we always have to tune them on site. We can’t pre-calculate it well enough because maybe the resonant frequency is correct, but the simple act of moving it into a room and placing it into a corner changes the tuning of the room. The other concern is the decay of the Helmholtz resonators. With all these active resonators you have to be really careful otherwise you end up having a resonator with a decay which is longer than the decay of the room. Then you hear the resonator, which we don’t want.” Veith doesn’t have a trademark name for his type of acoustic design, other than building it around the general concept of a Reflection Free Zone, his goal is to simply work with the room, the budget and the speakers. “I always say we have a toolbox with which we work; simulation programs, different absorbers and approaches,” he said. “We try to do the best for the room that we can in different situations. Maybe in one room it’s better to have free-float monitoring and another room we have soffit-mounted monitoring. I’m not totally fixed on one approach. I always have to find out what possibilities we have, the budget and a lot of factors, then I have to make my decisions to get the best results.” Veith was in the final tuning process when AT got a preview tour. Although Veith had just stepped off a long haul flight the day before, it was all smiles; he was very happy with the balance of the room when he first walked in. “I would say we are already 99% there. So now the fun begins to tweak and improve it even a little bit more.” There’s no magic trick to the final part, just lots of listening, measuring, listening and measuring. “You hear
things, then you find it in the measurement, and vice versa. Maybe you do something and it doesn’t sound right, so you go back. You always have to be very careful when making big changes.” Veith’s go-to listening CD is Back on the Block by Quincy Jones. “I’ve just heard it so many times, I’m so used to it. When I’m at the point where I’m happy, it’s like I feel at home. I use some for stereo image and some are for low end frequencies. I always want to change to a few new songs, but I have to find time to do it.” 301 REASONS
While Misner might not give them the credit, there are some other world-class facilities in Australia that happily cater for a coterie of international clientele. However, there’s no other studio in Australia like Studios 301 at the moment. The sheer number and variety of studios; the uncompromised acoustics; the impeccable PMC monitoring throughout; and a 200sqm-plus live room. Still, what does it say about the Australian music industry when it’s best recording studio is a charity case? Well, while Misner may have dropped a wad of cash building a superlative studio, the boots on the ground, daily grind will have little to do with him when he returns home. Spending the last couple of months watching the studio come together, there’s a palpable sense of relief in the air as some of Australia's best engineers and producers have finally been able to come ‘home’ again. After a year being split up all over Sydney, the team is finally reassembled and energised to get to work in rooms that will allow them to do their best work yet. Mastering engineers Steve Smart, Leon Zervos, Andrew Edgson, Ben Feggans and Harvey O’Sullivan are all settling into the sound of their purpose-built suites and PMC monitoring. Vocal producer extraordinaire Simon Cohen has staked out his turf in Studio Four. Head assistant engineer Owen Butcher is already busy running orchestral sessions. Steve Crane is installing the final touches, and GM Ron Haryanto and his team are busy introducing old friends to the new facilities. It’s not just about building a world-class studio; the aim is to provide six-star service and quality, whether the session is for one day or two months. They couldn’t have any better start than the new Studios 301.
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Despite progress made in combatting sexism, the numbers still don’t favour women in the music industry. An Instagram account called ‘Lineupswithoutmales’ appeared just prior to International Women’s Day and documented the imbalance by editing festival posters to leave only female-represented acts on the bill… some were left plain empty. DJ Mag’s 2017 readers’ poll collated a similar imbalance when its list of 100 most popular DJs featured exactly three female DJ acts. In recording studios, the ratio is similarly lopsided; female engineers, mixers and producers are estimated to fill only three percent of control room Herman Miller Aerons. Other awards haven’t fared much better. The recent #metoo movement famously fell on deaf ears during the 2018 Grammy Awards ceremony, as the results simply confirmed the skewed reality. Only one female artist, Alessia Cara, won a major Grammy Award (for Best New Artist), while a meagre 17 awards (out of a total of 86) went to women or female-fronted bands. This year’s British Music Producers Guild Awards had a different look. Five of its 16 MPG Awards went to women, including some of the most prestigious awards, like UK Producer of the Year, awarded to Australia’s own Catherine Marks (on the cover of Issue 94). In addition, Manon Grandjean received the Recording Engineer of the Year Award, and Marta Salogni the Breakthrough AT 22
Engineer of the Year Award. The other two were non-studio Awards, for Jane Third (A&R) and Imogen Heap (Inspiration).
especially women, tell me that seeing someone like me successfully navigating this industry makes them feel more confident in themselves.”
ALTERNATIVE POWER
GROUND MADE AT MPG
More than 25 years ago, I wrote an article about the dearth of women engineers in London studios. One studio owner remarked, ‘The rock ’n’ roll industry is one of the most racist and sexist industries in the world.’ The 2018 MPG Awards may be a sign biases are improving in Britain; Salogni’s own meteoric rise to the top of the London studio game seems to indicate as much. Yet Salogni says it wasn’t all laid out smoothly before her. “I always have to exceed expectations in the beginning,” she said. “People go, ‘That’s strange, a woman in this job.’ It means I have to do my job better than anyone, to justify the fact I am still a rarity. “Being a kind of exception is both a good and a bad feeling. Good because it feels empowering to be part of a ‘change’; I’m not only exercising a passion and a profession, but also providing an alternative to the status quo. Bad because it can feel isolating and uncertain, as I have very few role models to look up to. There are so few female engineers, mixers and producers in the industry that when I started my career there were moments I doubted my chances of succeeding. I speak to many young people now who are trying to understand how to make it in such an unpredictable industry. Lots of them,
For Salogni, the MPG Award was the icing on the cake after a particularly exciting and successful 2017, in which she, amongst other things, mixed most of the latest album by one of her favourite artists, Björk. Salogni mixed eight out of the 14 tracks on Utopia, and mixed vocals for two others. With New York-based Egyptian mastering engineer Heba Kadry mixing the other six tracks, and Mandy Parnell mastering, Utopia’s final stages were entirely delivered by female hands. It befits the theme of female empowerment; one of several subjects Björk explores on Utopia. Salogni wasn’t aware of Utopia’s subtext when she received a phone call from Björk in July 2017. “It was quite an incredible moment to suddenly have one of your favourite artists reach out to you,”
Artist: Björk Album: Utopia
I want to be in partnership with machines, rather than stay within their limits. I always try to make it do stuff it’s not designed to do
she said. “Björk can get whoever she wants to mix her albums, and the fact she chose to work with so many women on her new album is quite powerful. It certainly also helps to defeat the preconception that mixing or mastering is a man’s job, and will hopefully inspire young women today. The initial request I received was to mix two tracks from Björk’s new album, to see whether my style would fit. I did those two mixes at my studio, and the feedback was really positive, so Björk asked me to mix more songs for her album.” Salogni getting the call was a reflection of how far the Italian has come since moving to London in 2010, fresh out of secondary school. Salogni grew up in northern Italy, not far from Milan. As a teenager, she developed a keen interest in art in general, and music and sound engineering in particular: “I saw music and manipulating sound as similar to painting. It’s a form of fine art.” Salogni also took an early interest in synthesisers, which she calls “democratic,” because “to play a guitar, for example, one is traditionally supposed to know how chords work, what the finger positions are, and so on. Synths instead can be explored without any musical training, and with only the desire to create something new. Synths are intuitive and creative, you can sculpt away to find sounds that are unique.” WRENCHING ON TAPE MACHINES
After arriving in London, Salogni attended Alchemea Music Production College, which taught her the “rudiments of recording and Pro Tools,” and gave her experience on the college’s three consoles. Upon completing her course, Salogni got a job in the audio department of a movie post-production company, but realised she really wanted to work in recording studios and landed herself a job as assistant to producer Danton Supple. Initially working from Dean Street Studios (formerly owned
by Tony Visconti), Supple and Salogni went on to work at Strongroom Studios in East London. It was here that Salogni met one of her main mentors, star mixer David Wrench. She became Wrench’s assistant, and today Salogni still works with Wrench as an engineer “if projects and schedules align, but I now mainly mix and produce my own projects.” By the time Björk’s team contacted Salogni, the Italian had clocked up credits with Glass Animals, Goldfrapp, White Lies, and The XX. While popular acts, they all lean to a more experimental musical vision and sensibility. Salogni’s own taste skews to the psychedelic electronica of White Noise’s Electric Storm, and avant-garde album The Feedback by Il Gruppo di Improvvisazione Nuova Consonanza, which featured a young Ennio Morricone on trumpet. Salogni’s experimental mindset is exhibited in the way she wrangles the two-track tape recorders lining her room at Mute Studios in West London. “I have a weird relationship with gear,” explained Salogni. “I always try to make it do stuff it’s not designed to. I want to be in partnership with machines, rather than stay within their limits. For this reason I am fascinated with tape recorders. They may seem dated to modern musicians and engineers, but when I first encountered one a few years ago, I thought, ‘wow, this is beautiful!’ MAGNETIC RECORDING IS A BEAUTIFUL
TR303, Roland System 100, Sequential Circuits Pro One, System 100 and more. All the synths are hardwired into an an SSL Matrix desk, which is the heart of Salogni’s mixing setup, alongside a Pro Tools system, Dynaudio BM15 and Adam A7 monitors, and a Pure consumer radio.
CONCEPT, AND I DO ALL SORTS OF THINGS WITH THESE TAPE
OBVIOUS HOW TO ORDER THE TRACKS, BECAUSE THERE WAS
RECORDERS, LIKE USE THEM AS DELAY UNITS, OR TO CREATE
NO TRADITIONAL BAND STRUCTURE TO INDICATE WHAT
POLYRHYTHMS, SATURATION, DISTORTION, AND FEEDBACK.
THE INSTRUMENTS ARE AND ROUGHLY WHAT THEY HAVE TO
Given that they were purely built to record, I think that’s pretty good! It’s why I have several of them: the Revox PR99 MK III, Akai 4000DS, Ferrograph 5A and a Davoli Echo Mixer. I’m completely in love with them!” Salogni’s studio also contains quite a bit of analogue outboard and a collection of synths and drum machines that belongs to Mute owner Daniel Miller, including a MiniMoog, Polyvox, Roland
SOUND LIKE.
STEM ORIGINS UNKNOWN
With just stems to work with, Salogni didn’t have much cause to inject her tape machines and synths into the mix. She simply dove right into the ’Tools session. “I started almost the moment I put down the phone,” she said. “They sent me the Pro Tools sessions for The Gate and Arisen My Senses. The session for The Gate contained only 13 stems, so there was a minimal amount of tracks. As my mix developed, I split the main vocal stem and the main processed stem to be able to treat them differently for different moments of the song. Some additional stems arrived from Björk, giving me a total of 16 stems to work with. “Normally when I start mixing a track, I spend an hour colour coding and naming, then I order the tracks with the drums at the top, the vocals at the bottom and the instruments in the middle. Within that structure I also place the tracks in the order they appear in the timeline. You must have some kind of method! IN THIS CASE IT WAS NOT
“You know what a kick drum has to sound like, but here there would just be percussive tracks with high end and low end. Also, normally I go through the tracks one by one and EQ things, but in this case all the elements are interconnected. I could not go in and EQ individual tracks, and expect it to sound better. Some of the stereo stems included a wide range of elements, and therefore frequencies, AT 23
THE GATE MIX As you might expect from a Björk track, the stems Salogni received for The Gate included esoteric-sounding jobbies like ‘Extra Flute Arpeggios’, three tracks of ‘Air’, and ‘White Noise’. Once Salogni added her five effect aux tracks and a master track, the entire session came to a modest total of 23 tracks.
IN THE BOX Salogni: “I normally use outboard when I mix, but for the Björk mixes I kept everything in the box, because I wasn’t sure whether I had to do recalls, so it was indeed all plug-ins and Pro Tools
VOCALS Salogni’s treatment of Bjork’s lead vocal in The Gate illustrates her tendency to use many EQs in series on the inserts. The insert chain consists of two EQ3 7-bands, a CLA-76, a ProQ2, a C4 multiband, and a Waves RDeEsser, cutting 5500Hz. “All the EQs are subtractive, with the first having a hi-pass at 78Hz, and cutting at 98Hz, 175Hz, 290Hz, and 740Hz, and the second one cutting at 2.75kHz, the latter because the vocal got harsh in a few places. The CLA76 doesn’t do too much, because I didn’t want to mess with the dynamics and my intense volume automation. The Q2 does more notching.”
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automation. With the latter not only volume but also panning. Sometimes I think I’m the only one in the world who likes the Panman, but it can be a bit random, like the Waves Mondomod, which I also
occasionally use. If I want be precise, I do panning automation by hand in Pro Tools.”
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LIMITED TOOLBOX Although Salogni repeatedly stressed the importance of volume automation in her mix, she used a respectable amount of plug-ins. What is striking, is that she seems to use many different instances of a select few plug-ins, mostly the Avid EQ3 7-band, FabFilter Pro-Q2 EQ, Valhalla Vintage Verb, and a few Waves plug-ins: the C4 multi-band compressor, CLA-76 compressor, DBX160 compressor, HDelay, Linear Phase Multi-band, and DeEsser. The master bus has a couple of different plug-ins, namely the Waves G-Master Bus compressor and the PSP
Master Q2 EQ, plus the aforementioned Linear Phase Multi-band. In other sessions she uses a few additional plug-ins, like the SoundToys MicroShift, Little AlterBoy, and Panman, and some of the Air plug-ins, like the Phaser. But that is it, more or less. With most mixers swimming in plug-in lists numbering in the hundreds, this is fairly unusual. “I don’t need any more plug-ins,” Salogni explained. “I believe in limiting yourself to a few plug-ins, and then exploiting them as much as you can. Otherwise I’d feel lost with too many choices. It’s like having
a pedal-board that’s too big! 99.9% of my delays come from the HDelay, often with the ping-pong delay for stereo. I like the Valhalla plug-ins for reverb, usually the Vintage Verb. I very occasionally use the UAD 140 plate. When I want to give some more space to electronic drums in particular and want them to sound a bit more organic, I use the UAD Ocean Way Studios re-amping plug-in. I tend to use the EQ3 7-band and the Q2 for EQ because in general I don’t use EQ to add character. If I do want colour I may use the UAD Pultec HLF-3C.”
MASTER BUS Salogni: “My master chain usually begins with an EQ, but if I feel it doesn’t need that, I’ll start with a compressor. In this case it was the Waves SSL GMaster Bus, which adds very subtle compression with a ratio of four. Then there’s the PSP Q2, gently notching out a few frequencies — 202, 345, 495, 624Hz and 2.32kHz — and then I have the Linear Phase Multiband. After that I’ll have the Waves L3LL Multimaximizer, for some rough simulated mastering, to be able to hear the things at a louder volume. I take the L3 off when I send the tracks to a mastering engineer, but when I finalise the mix I need to make it louder to be able to understand what it will feel like once it is mastered. Otherwise it may come back from mastering sounding completely different.”
MIXING VOLUMES Salogni: “Different frequencies sound different at different volumes. This is also the reason why I mix at medium volume. I’ll then occasionally turn it up for a few minutes, to check the mix, and I do the same with a lower volume. I want to make sure
EFFECT TREATMENTS Salogni: “I actually split the vocals over three tracks, for the intro, verse and chorus, to be able to treat each differently. Each of these tracks also had different sends to my aux tracks. One aux was a delay aux with the Waves HDelay on my usual setting; a ping pong with lo-fi, but no analogue noise. I also had a Reverb aux, which had the Valhalla Vintage reverb, which is my favourite; all I do is change the pre-delay, the decay, the low-cut so I don’t mud-up stuff, and the hi-cut depending on how hissy I want it to be. There’s an Aux with the Valhalla Shimmer, but I use that just for one moment, because it’s become so recognisable now. I put a Pro-Q2 after the Shimmer, because it can sound pretty harsh, taking out 1kHz, another frequency I’m not particularly a fan of. The verse vocal only had a send to the Valhalla Vintage reverb aux, while the chorus vocal has sends to the Vintage Reverb aux as well as the Delay aux with the ping pong HDelay, for greater width.”
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that the mix sounds good both quiet and loud. The songs are coming from a very different aesthetic than the one of a common radio mix. The challenge was to make the songs as powerful as a modern mix but still to retain maximum dynamics. EQ,
multi-band compression, and automation helped a lot to achieve this. My goal was to create a sense of complete immersion in the song for the listeners, to create sonic shapes that create a unique sound world for each song.”
AT 27
I saw music and manipulating sound as similar to painting. It’s a form of fine art
hence my use of multi-band compression. All elements on the stems are interconnected and need to be addressed as a whole entity, rather than individual parts. A big part of my process was working out what was in the stems, and how those things fit together musically; then, to find a way of organising my mix session and a way of mixing that worked.” LOST IN THE ROUGH
The Gate eventually became the lead single of Utopia, something Salogni was not aware of while mixing. It’s a very spacious track, in free rhythm, with a backing of flutes and electronic sounds, a deep 808-like bass, some non-rhythmic percussion, and quite a number of backing vocals. Given Björk and her team, which includes co-beatmaker and co-producer Arca, had already managed to whittle down the mix to 16 heavily-treated stems, one wonders what they hoped Salogni would add. It was her question too, not made any easier by the lack of additional information or direction. “I always start a mix from what I’m given because I assume that’s what the artist is comfortable with,” she explained. “The artist and producer will have been listening to the rough mix, and that’s what they’re keen on. My job is to translate that into something better. However, with these two songs they purposefully didn’t give me the reference mixes. I asked for them, and Björk said: ‘I want to see what you bring to this, completely from your own imagination.’ That was empowering and intimidating at the same time! I had no idea what their starting point was. Obviously I had to start from what was in the session, but they might already have created a killer mix, and I’d have absolutely no idea what that sounded like!” Still, Salogni pressed on: “I was not going to add plug-ins to sonically transform the stems, because they already had such unique and definite sounds. Instead my main aims were to create clarity and let the dynamics flourish. In creating this track they had already thought a lot about the dynamics. I wanted to enhance them and make them much more extreme. I wanted to make the track euphoric in a way that really grabs attention, and have things jump out in the mix. There were only 16 tracks, but it still meant a lot of detailed volume automation that took a long time to do. “Another aspect of the dynamics was to make AT 28
things more spatial, width-wise. I am pretty bored with regular stereo. I like widening and making mixes as immersive and three-dimensional as they can be, while at the same time being respectful of the space that elements are meant to have. During subsequent mixes Björk gave me insights that allowed me to maintain the natural stereo image placements of the choir and flutes, in accordance with their positioning in the church or hall where they were recorded. However, for the first two tracks I mixed in London I had no idea and had to go with my instincts. I think that’s also why I was given stems, so I had no option but to preserve the panning! “WITH REGARDS TO CLARITY, I ALWAYS DO A LOT OF SUBTRACTIVE EQ, APPLYING IT IMMEDIATELY AFTER I HAVE ORGANISED THE SESSION. Unnecessary low frequencies take a lot more space and can create a lot of mud. I use two EQs for this — the Avid EQ3 7-band and the FabFilter Pro-Q2, because they are the most neutral-sounding. If there’s still a frequency that annoys me, I’ll use another Q2 to sweep the frequency range and take it down. I then bypass the plug-ins to see if I’ve made an improvement. In general I am not a fan of 200Hz, 400Hz, and 600Hz, basically all the harmonics of 200Hz. I also don’t like 3kHz and 6kHz. The beauty of creating space by taking out frequencies is that you then have space to boost one thing. In general I like to use several different EQs in series. It’s a matter of mixing up different colours.”
MIXING IN BJÖRK’S BACKYARD
Salogni stressed that her overall aim when mixing The Gate was “to polish what they had done, to give it justice, to keep some elements on top of others, and also make sure everything felt like it belonged in the same world. The vocals had to be intelligible, but also not overpower the instrumentation. The low end was really important, and I wanted to give that as much impact as possible, while ensuring the high end was not piercing.” Salogni was clearly successful, because “after I sent my two mixes to Iceland, the feedback was really positive, and the changes I was asked to do were minimal. That felt amazing, to have been able to interpret Björk’s vision. When Björk then asked me to mix more tracks, she said, ‘should I come to London, or would you like to come here to mix?’ There was a time constraint in that she could be in London for less time than I could be in Iceland,
so I suggested I head over. There’s also an obvious connection between her music and the Icelandic landscape and culture, plus she could stay close to her family.” Björk and Salogni worked on the mixes for Utopia in Reykjavík in a new studio complex that belongs to engineer and trombone player Bergur Þórisson. “We were set to go to a studio that was fully equipped and sound insulated, but it wasn’t available, so Bergur offered his studio. The room wasn’t finished yet, and not sound insulated, which meant it was quite hard to work in because I’m used to super-dead spaces without leakage. It was challenging, but I actually loved it, because it really pushed me to my limits. Björk and I really wanted to make sure the songs would sound good on any system in any room, so I asked for as many speakers as I could. We listened to Auratone 5Cs, Adam A7s, and Amphion Two18s with an Amphion BaseOne25 subwoofer. We also listened in Bjork’s car, on her Genelecs at her house, and on Beyerdynamics DT770 headphones.” Salogni mixed another six tracks while she was in Reykjavík, plus vocal mixed two more. Overall, her mix approach for the tracks was similar to that of The Gate, but sitting in the same room with Björk during this process proved a big help. “She would give me a lot of background information about the recordings and song concepts, which was very important to me. She was really good at communicating what she wanted, often combining words, gestures and images. If something still wasn’t clear to me, she would just show me in Pro Tools what she had in mind. She’s really good at Pro Tools, and that was also very helpful. “While I was in Reykjavík I did further tweaks to The Gate and Arisen My Senses, and another thing that made a difference was getting to see the video for The Gate. That changed my perspective. It helped me not only understand the feeling she’d explained to me, but it also clicked visually. I went in and did some more automation inspired by the video. For example, the moment the low frequency sound comes in, she makes a movement with her hands and I wanted to replicate that in the mix. I WANTED THAT MOMENT TO BE AS IMPACTFUL AS POSSIBLE, AND HAD THE TRIM AUTOMATION UP BY 4DB OR MORE. I love music videos that have a strong connection with the music. Working in post-production, whenever the visual cues and music cues matched, it created another level, a fourth dimension.”
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TUTORIAL
VIEW FROM THE BENCH
When 300Hz is ruining your mix, these are the types of EQ that could save the day. Column: Andy Szikla
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‘Three Hundred Hertz!’ was all I could hear over comms, people were shouting it from every station, ‘Three Hundred Hertz!’ I was spread-eagled across every suspect fader on the mixing desk, whilst feverishly poking at the 31-band graphic equaliser trying to backtrack the mysterious hum that had just trodden all over my mix. Nothing worked, and the hum just got louder. It was a big gig for me at the time. A Fortune 500 corporate breakfast with a fully stocked crew, and a show director who was now glaring at me, wearing the I’m-upset-and-you’ll-never-work-again face. Having run out of ideas, I stood up, walked over to the nearest loudspeaker, and stuck my head in front. There was no hum. So I followed the sound through a doorway that led to some offices, where I discovered the janitor hard at work with a big noisy vacuum cleaner on his back. It was 8:30am. But wait, there’s more. At exactly the same time, a street-sweeping truck was working its way up the rear lane behind the venue, the sound of its rotary brush and vacuum, increasing in volume as it drew closer. Wooooo. I lumbered back to my post and spent the next five minutes restoring all the settings I messed up during the scare. The moral of the story; EQ is so well-known, people will shout frequencies at you even when the things that go ‘Wooooo’ have nothing to do with you. But how much do we really know about what’s going on behind those dials and faders?
EEK! WOOOO!
EQ is short for Equalisation, which in turn is short for Frequency Equalisation, which is fancy talk for Tone Control. It sounds about as simple as primitive tone controls actually were. Modern EQ, on the other hand, can get very fancy indeed. EQ enables a sound technician to apply corrections to room acoustics and loudspeaker non-linearities, while also enabling their own aesthetic and artistic choices in both sound reinforcement and recording. FREQUENCY THROUGH TIME
In 1892 Alexander Graham Bell’s telephone system had connected New York to Chicago with a pair of wires over 1400km long. The two parallel wires effectively functioned as capacitor plates, and pretty soon it got up everyone’s nose how the whole thing behaved as a geographically-sized, high-cut filter. To make up for the resultant loss of sibilance and intelligibility, Bell Laboratories developed and installed fixed-frequency, non-adjustable filters to restore the top end, and thus ‘equalise’ all frequencies in the telephone signal. The first user-adjustable equaliser is credited to John Volkmann from RCA, who created one for movie playback systems in the 1930s. Most picture theatres of the day were built in the silent era, or were converted song and dance venues. The ‘talkies’ brought electronic audio into those venues for the first time. Between the primitive speaker
systems, the sound recordings themselves, and the lucky dip of theatre acoustics, it often added up to an unsatisfactory day out for the ticket-paying punter. Volkmann’s gadget offered boost and cut of multiple selectable frequencies, and to theatre managers the altering effect must have seemed almost magical. At the same time, other companies were developing equalisers to be used in audio production and broadcast. Art Davis of the Cinema Engineering Company hit an early home run by designing the first proper graphic equaliser, the Type 7080. Featuring six bands with 1.5 octave spacing, the unit offered 8dB of boost and cut, zero insertion loss, and faders fashioned from linear 17-position switches sporting ‘typewriter key’ knobs. The unit was an early active device, which employed variable passive filters to alter the cathode resistance of a tube amplifier, and therefore its gain. At the back end there was some make-up gain followed by a push-pull section, driving a transformer that delivered a floating differential output. It was clever stuff for its day. Later, Art Davis defected to Altec, where he designed the seven band Model 9062A graphic equaliser, which remained a benchmark device till the 1970s. In 1952, Englishman Peter J. Baxandall published his scheme for a negative-feedback tone control circuit using potentiometers as opposed to switches, thus allowing full user control. It also ushered in a new era of fully-active EQ circuitry AT 31
INDUCTED INTO OBSCURITY
Looking at the work of Massenberg and his contemporaries, one aspect that never gets talked about is how they stood astride a historical line of demarcation for the electrical component called an Inductor. Before Massenberg, if you took the lid off an equaliser you would probably see inductors. After Massenberg, probably not. Inductors and capacitors are components with electrical properties that vary with frequency, and they are what shapes the response of any frequency or time dependent circuit. They work in equal and opposite ways, so it was always natural to see them together in the same devices, the EMI REDD desk EQ from the ’60s being one fine example. However, inductors are essentially a coil of wire with properties similar to a radio antenna. In Massenberg’s paper, he described them as marginal AT 32
FIXED EQ 40dB
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and became the most reproduced tone control in history. To this day there has scarcely been a hi-fi unit anywhere in the world which fails to incorporate it behind those knobs marked Bass and Treble. Baxandall received no royalties for his invention, but in 1950 the British Sound Recording Association gave him a $25 gold watch, in honour of an earlier version. That watch would be worth something now. Meanwhile in Texas, Professor Wayne Rudmose had figured out that public address systems needed more than just a bass and treble control to cure feedback. His 1958 paper Equalization of Sound Systems proposed that PA systems could actually be tested for frequency response (using non-existent equipment that would need to be devised), and thereafter filters might be created and installed to equalise that response. His theories were successfully put into practice at Dallas Love Field Airport, and from then on ‘tuning’ a PA became a thing to do. The next decade saw a lot of work done to further tame PA systems. Rock ’n’ roll was in full bloom, and program material was getting louder. In 1967, Altec Lansing introduced the first 1/3-octave passive notch filter set specifically for PA tuning called the Acousta-Voice system, and from there it was only a hop and a jump to create a 1/3-octave, cut and boost-style graphic equaliser. The last truly major development in EQ was the parametric equaliser. Historically, graphic equalisers were useful studio tools, but had big faders, and took up lots of space. What engineers craved was a smaller and more flexible device which could be built right into a channel strip. In 1971, Daniel Flickinger invented a sweepable EQ which allowed adjustment of frequency and gain, in three overlapping bands of fixed width. Nowadays, manufacturing a mixing desk without at least one band of sweepable EQ is unthinkable. In 1972, George Massenberg presented a paper on a similar invention that he and two associates had been developing which, besides frequency and gain, allowed user adjustment of the third parameter — bandwidth. Massenberg called his device a ‘Parametric Equalizer’ and he continues to sell them to this day, under the GML brand name.
-24dB 100Hz
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performance components which inherently invite parasitic electromagnetic noise into a system, and he proposed the use of several pre-existing electronic Indian rope tricks to do away with them completely. One was an op-amp circuit called a ‘gyrator’, which takes a capacitor and, more or less, draws a moustache and glasses on it to fool the neighbouring electronics into thinking it’s an inductor. The ideas he talked about increasingly appeared in the work of others, and gyrators soon replaced inductors in graphic equalisers. Since then, inductors have become a sad and sorry underclass in audio, and now you only really encounter them inside passive speaker crossovers. If you come across one, loan it a dime. ALL THINGS BEING EQUAL
Rudmose used fixed frequency notch filters to equalise the PA in Dallas, and that was that, no knobs. These days fixed EQ is less common, but
the RIAA curve has been a solid stayer. On a vinyl record, the grooves wiggle from side to side, and the bass notes (which contain the most energy) wiggle the widest. So in order to fit a useful amount of material on the vinyl the bass is attenuated by 40dB at 20Hz, rising to unity at 20kHz. The filter slope zig-zags between those two points, but what you hear is closer to a straight line. All records are made that way, so a phono input employs a fixed boost filter of the opposite slope, to put everything back to normal. Most mixing desk channels feature a sweepable or switchable high-pass filter to attenuate sub-bass and proximity issues in microphones. Occasionally you will see a sweepable low-pass filter, to help remove unwanted treble from bass instrument channels. These filters provide a flat response in their pass band, and generally roll off down a 6dB or 12dB per octave slope beyond their cutoff frequency. Filters of a higher order than 12dB
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BANDPASS FILTERS
per octave are less common, and tend to imply a special purpose. Linkwitz and Riley famously use 24dB per octave filter slopes in their electronic speaker crossovers. If you want to affect a specific band of frequencies, you need a bandpass filter. The frequency response is usually bell-shaped, with an upper and lower roll-off slope, and there are two main varieties — Constant Q and Variable Q. ‘Q’ stands for Quality Factor, and is simply the ratio of -3dB bandwidth in Hertz to a centre peak frequency. For example, if we have a centre frequency of 1000Hz with -3dB roll-offs at 414Hz and 2414Hz, then the bandwidth (upper minus lower) equals 2000 Hz, therefore Q=0.5 (1000/2000). For amplitudes less than 3dB there is no -3dB point, and subsequently all kinds of disagreement about how to calculate bandwidth. I don’t want to start a bar room fight, so I will stick with the above as a general principle, and just say that whenever we increase Q, we decrease bandwidth, resulting in sharper curves. Bandwidth can also be measured in octaves, which are very roughly the inverse of Q — but always slightly more, because they are calculated from either extremity rather than the centre. Since an octave is merely a doubling of frequency, the distance between 414Hz and 2414Hz winds up being about 2.5 octaves.
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When boost or cut is applied to a bandpass filter, bandwidth and Q will either be constant or variable, and the term ‘variable’ does not mean adjustable. The frequency response curves for a variable Q equaliser show that the bandwidth or Q of the equaliser changes with the amount of boost or cut used. Sometimes this is called ‘Proportional Q’. At low settings the equaliser exhibits a wider relative bandwidth than at high settings, and only achieves its specified bandwidth at maximum boost or cut, so minor adjustments made at a specified frequency will cause changes to other frequencies some distance away. This type of response is desirable in situations where you want to change the overall tonal balance of a sound, without overemphasising any particular frequency. Variable Q circuitry is common in old-fashioned tone controls, guitar amps, and most professional equalisers that fail to declare themselves as being ‘Constant Q’. The frequency response curves for a constant Q equaliser show that, even for small amounts of boost or cut, the equaliser maintains its stated bandwidth. That makes it very useful for applying changes to specific frequencies, in circumstances where you would prefer to leave nearby tones undisturbed. Examples might include removal of feedback, or enhancement of a particular resonance in a musical instrument. GRAPHIC EQ
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10kHz
In live sound reinforcement, the 31-band graphic equaliser has left its footprint more than any other EQ. With evenly spaced 1/3-octave bands, it gets its name from the way the response curve is so directly displayed (like a graph) by the faders that
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+15dB
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GOING OVERBOARD FOR OUTBOARD I talked to Kaj Dahlstrom from Melbourne’s renowned Sing Sing Studios, and asked him which outboard EQ units have been most popular with his customers, and he pointed to the following units: • George Massenberg Labs GML 8200 five-band parametric EQ is extremely popular amongst seasoned veterans who want to do serious tone shaping. Kaj notes that users like the sound it makes when air frequencies are boosted. • Kaj says most like to track using the Neve 1073 (or 33115) which offers a high and low shelf, and a switch-select sweepable mid with fixed Q cut-and-boost. The design is all discrete transistor, and old units with the original Marinair input and output transformers are highly prized.
control it. When you are battling wild frequencies in a pesky room, it’s helpful to quickly see where you’ve been. Live sound techs talk about their equipment the way a tradesperson talks about their ute. ‘I only nudged the fader, but it produced enough grunt to uproot this tree.’ In that context a good graphic equaliser will be one for which each fader exhibits a solid grasp of its band of frequencies, so that when you move one ‘something happens’ (I have used equipment where nothing happens). Graphic equalisers mainly employ constant Q equalisation, so that bands several faders apart from one other don’t interact, but a superior design will allow bands immediately adjacent to ‘combine’ slightly when the faders are moved together, eliminating any dip that might occur between them. Combining was invented by Art Davis when he created his Acousta-Voice system. He reasoned that if your P.A. is feeding back at a frequency half-way between two adjacent bands, then you will attenuate both, with the expectation that all frequencies between the two ought to be grabbed in the process. PARAMETRIC EQ
In recording studios, room sound and feedback are not quite as high on the worry list, and AT 36
• Focusrite ISA115 is also popular, and has a mic pre which people like. It was originally designed by Rupert Neve as a channel strip for the consoles at George Martin’s Air Studios. It functions a bit like a Neve 1073 with an extra mid, and features Lundahl transformers in and out. • Chandler Curve Bender is somewhat revered, and gets used a lot on big-time vocals. It is a re-creation of the EQ section from a channel of the TG12345 desk made by EMI, which was the console that replaced the valve-based REDD desks of the Beatles era, and was used on Pink Floyd’s Dark Side of the Moon. Units provide two channels of four-band sweepable EQ with cut-and-boost, as well as a high-pass and a low-pass filter. Old school tapped inductors are used in the EQ sections, and user adjustment is facilitated using stepped switches instead of potentiometers.
EQ tends to be more purely applied to affect the timbre or balance of an instrument. With independent control over bandwidth (Q), frequency and amplitude, a parametric equaliser is able to produce almost any shape of EQ curve one might desire — something the fixed frequency bands of graphic EQ cannot do. Circuits tend to employ a state-variable filter topology using opamp integrated circuits, and I have never heard of a parametric EQ using valves. BAXANDALL TONE CONTROL
This classic tone control provides a shelf, as opposed to a peak-type response. A stock standard configuration might produce 15dB cut and boost at 100Hz and 10kHz, with the slopes returning to unity not far either side of 1kHz, but there are countless variations. VIRTUAL
In this day and age of software-based recording, there are any number of proprietary algorithms that exist to mimic all of the above, and more. I’m with the peeps who favour minimal EQ and careful mic placement during tracking, though for mixing I will admit to being quite a fan of software equalisers. There are many who love to mix or print through analogue EQ equipment. In the end, if one way sounds better than another, then it is.
EQ PHONE HOME
All that’s useful, of course, but Mrs Tech Bench asked me if there’s a way to manage your woofs and tweets without having to think about it. Well... I tell her about a junior conference technician I knew named Mark. Mark is a genius. On his first real test, he is left alone with a PA and a medical professor on stage, who blathers into his lapel mic about the exciting world of cortico-steroids. Mark thinks he can hear the onset of feedback, but what to do about it? He pulls out his phone and calls his colleague Dave, who is famous for being able to identify any feedback frequency instantly. Mark sings in a whisper down the phone line, ‘Wooooo’. ‘Three Hundred Hertz,’ says Dave. Mark reaches over to the graphic equaliser, pulls down the 300Hz fader, and the problem is gone. ‘Thanks Dave.’ ...Now that’s the way to EQ.
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REGULARS
PC Audio Nostalgia isn’t what it used to be — now it’s all about software emulation! Column: Martin Walker
My time with PC Audio started way back in the days of 8-bit computer games, when clients started to ask if I could compose music for their PC games, as well as for the Commodore 64, Amiga 500, and Atari ST machines I was already working with. At best, those early PCs could manage a beep, courtesy of a tiny buzzer on its motherboard whose voltage could be toggled high or low. So, for instance, if you wanted to produce an A440 squarewave (the only possible waveform!), you had to code a routine that sent the appropriate commands 440 times a second. Some PCs even had the luxury of a tiny loudspeaker (typically with the same capabilities, but at least capable of making a somewhat louder noise). Then the industry moved on to the first plug-in soundcards, from the humble Adlib (with rather tinny 11-voice FM synthesis courtesy of the same Yamaha chip used in its PSR and PSS keyboard series), to the first Soundblaster soundcard, with 8-bit audio sampling. I even got to write MIDI-based music for Roland’s LAPC-1 PC expansion card! CHIP OFF THE OLD BLOCK
Just recently I had the opportunity to dust off my extremely rusty programming skills, when I was once again asked to write some music for the Commodore 64’s famous SID chip (Sound Interface Device). The C64 was the best-selling home computer in history (apparently over 17 million were sold), and SID was also the most famous 8-bit sound-making device of the time. Devised by engineer Bob Yannes (who later cofounded Ensoniq), it contained three oscillators each offering four different waveforms, plus three ADSR envelope generators, a multi-mode filter, and even ring mod and sync options, all with hardware personality. This was revolutionary stuff in 1981! However, what really surprised me all these years on, is that the PC can now accurately emulate the SID chip via software. So, I didn’t even need a C64 to write this music. This time I could create it all on the PC, even down to auditioning the small changes in how this music would sound when played back by the various SID chip revisions manufactured over the years. Even better, the entire C64 is now available in PC emulated form for free download, courtesy of applications such as VICE (Versatile Commodore Emulator, available at AT 38
vice-emu.sourceforge.net). Frankly, I was amazed at how accurate these emulations were, even down to incorporating bugs in the original hardware that still need to be carefully worked around (like the infamous SID ADSR bug that occasionally results in a 33ms delay before the envelope switches between a slow and fast envelope rate). The SID chip has even been re-created for the PC in VST Instrument form, most notably by Plogue, whose chipsounds VSTi (plogue.com/products/chipsounds) authentically emulates 15 different vintage 8-bit era sound chips in incredible archaic detail. WHAT GOES AROUND COMES AROUND
However, as I waded back into 6502 assembly language programming to update my C64 music driver routine (this time assembled and run using my PC as a ‘pretend’ C64), I began to realise just how many parallels there are between coding and music-making in general. There I was, shuffling the order of my audio driver code segments to optimise their average CPU overheads, paying particular attention to avoid any unexpected peaks of demand that could result in the game graphics glitching or juddering. Then it suddenly struck me that this is exactly the same problem that still affects so many modern PC musicians — attempting to achieve low latency audio recording and playback without glitching, and for exactly the same reason. It only takes one errant Windows routine (even something seemingly unrelated, such as network sharing, or automatic updates) to unexpectedly demand more than its fair share of your PC resources to tip your audio buffers over the edge, resulting in audio clicks, glitches and eventually sonic juddering. Countless articles have been written (some by me!) on how best to tweak the various Windows versions to achieve lowest audio latency, particularly vital while recording with plug-in effects or playing back VST instruments in ‘real-time’. Modern developers also have to bear average/ peak CPU requirements in mind during the design of software plug-ins and instruments, as a single unexpected CPU peak demand significantly beyond the norm might stop your DAW in its tracks. Some developers may even offer run-time quality options in their products that manage the average CPU, to keep this overhead lower during mixdown when you demand ’real-time’
performance, but let you increase it to maximum when you render the final track off-line, to achieve higher audio quality. PEAKS & TROUGHS
The more I thought about this process of keeping an eye on the average and avoiding unexpected peaks, the more audio parallels I came up with. The most obvious one is compression itself, a technique designed to do exactly this — adjust the average audio levels while taming errant peaks — although we don’t want to be too heavy-handed about it and suck the life out of the music. Another common problem that so many of us face when mixing is allowing enough room for every sound to breathe. Once again, if we pile up too many similarsounding instruments at the same time we can end up with a muddy mix. Some resort to EQ sculpting on each track, to carve out some extra space for each element and let them breathe, but often a far more successful ploy is to consider the musical arrangement itself. For example, do you really need those six guitars playing at the same time? Does that bass sound complement the kick drum, or do they conflict frequency-wise? It’s often beneficial to juggle musical segments so that each one has its chance to shine in the overall mix, rather than fighting each other for space, exactly as I was doing with my code segments. There are, of course, negative ways to manage this balance between average and peak levels, as we know so well from the so-called loudness wars on commercial releases. Nevertheless, the quality of these gaming computer emulations running inside my PC via free downloads was stunning, and I realised how closely this parallels the audio plug-in world. There will always be a place for hands-on hardware (I still build some myself), but there’s now little discernable difference between the best plug-ins and the hardware they model. Yes, some will be able to tell the difference, but in a blind test I’d be surprised if many people could reliably tell which was which, and even those who could, might begrudgingly agree that using the plug-ins makes far more sense on the many occasions when a mix may need to be recreated in the minutest detail months after the event. Nostalgia isn’t what it used to be — now it’s far more about software emulation!
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REGULARS
Apple Notes What does Apple’s move towards ARM chips mean for Intel, Hackintoshes and the gradual assimilation of iOS and macOS? Column: Brad Watts
Well the cat’s out of the bag. As per my previous suggestion; Apple ditching third party-manufactured CPUs is now looking very likely. Last issue we looked over the ARM-based T2 chip in the iMac Pro, a processor designed by Apple to take care of encryption and SSD control, face recognition, startup security and access to the startup drive, and indeed, audio control. With the T2 taking care of processor intensive chores like encryption, there’s greater headroom for the primary CPU. According to every tech media body on the planet, and reportedly “those familiar with Apple”, the tech giant has a roadmap to move away from Intel processors and commence releasing both laptop and desktop machines using ARM CPUs. This is widely rumoured to begin in 2020. Intel shares took a nosedive, of course, dropping by around nine percent upon the news, with Apple’s business amounting to about five percent of the chip manufacturer’s annual revenue. Obviously, Apple has plenty of runs on the board with ARM processors. Since iPhone version one, released in 2008 with its paltry 412MHz ARM CPU, Apple has created an entire ecosystem with iOS. In recent years we’ve seen macOS (formally OS X) absorb various facets of the iOS system and vice-versa. Now, with the A11 ARM processor in the iPhone 8 and X models eclipsing MacBook i5 and i7 processing speeds, it makes sense Apple would be aiming for across-the-board use of ARM as its primary CPU. ARMED & DANGEROUS
While there are physical advantages to the ARM platform, such as far less heat generation, the advantages for Apple are most importantly; the freedom to release products according to its own development timeline, and to integrate software more closely with a bespoke CPU. But aside from this, there’s also the question as to when ARM processing will reach the performance required by workstation systems such as the upper end iMacs and the fabled 2018 release of a new Mac Pro. The A11 64-bit ARMv8 gets close. The A11 Bionic is a six core processor, with two cores running at AT 40
2.39GHz and the remaining four kicking in when required. It’s the fastest processor available in a phone to date, and as mentioned, outpaces various laptop designs from Apple. Manufacturing these chips is Taiwan Semiconductor Manufacturing Company (TSMC), a company that’s gone from US$1.5 billion to around US$25 billion during 1997 through to 2014. TSMC is a manufacturing powerhouse with a hugely successful history. With its prowess backing Apple’s charge, we’re likely going to see ARM processors reach the 3GHz barrier soon. Apple is certainly unafraid of a processor change. The company has done this before with the jump from Motorola 68040 architecture to the Power PC CPU, and then to the Intel processors such as the i5, i7, and Xeon platforms. As much as it pains the consumer, Apple have got these transitions down to a fine art. MAC PROS & CONS
Which leads me to speculation as to the 2017 announcement of a revamped Mac Pro release during 2018. It’s not happening. Recent news from Apple has set the 2018 Mac Pro farther out to 2019. Tom Boger, Apple’s Senior Director of Mac Hardware made the announcement, sighting the fact customers’ buying decisions would be greatly affected by the news. Damn straight — those customers have been making do with the 2013 Mac Pro, heavily modifying older Mac Pros, building a high-spec Hackintosh, or simply jumping ship to Windows. A 2019 Mac Pro announcement could easily equate to late-2019 in Apple-speak, with the actual release of a product being early 2020. The big question is; will the next Mac Pro use ARM processors? There’s also the question surrounding just how far iOS and macOS integration will become. All Mac users know professional workstation activities are best done with a desktop or laptop machine, with iOS applications amounting to watered down versions of their desktop counterparts. Notwithstanding, stripping Intel processors out of the software development map would allow Apple a single code-base; streamlining further
development. Who knows what new features will appear, or what pain it might cause developers. RIP HACKINTOSH
This will, of course, greatly affect the Hackintosh fraternity. It’s a dubious avenue at best considering Apple’s licensing fine-print, and is only possible due to the fact Apple uses Intel CPUs. I highly doubt an Intel processor could emulate an ARM CPU effectively, there’s next to no chance of motherboards being released to accept Apple’s ARM CPUs, and zero chance of buying an Appledesigned CPU on the open market. In other words it’s curtains for the Hackintosh, although the final nail won’t be driven in until Apple ceases support entirely for Intel-based machines. There’s a few years left in the DIY Mac yet, but the writing is unmistakably written on the wall. Change is afoot. This is speculation, and there are other issues at hand such as virtualisation on ARM processors. Running virtual machines is something Intel x86 processors excel at, whereas ARM does not. Maybe Intel processors will remain in high-end Macs. Maybe macOS will evolve to a point where it won’t run without an ARM processor present, but will still execute the hard work on an Intel CPU. Where this path leads, however, will be far clearer once we see the next Mac Pro.
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REVIEW
SONY C100, ECM 100U & 100N Hi-Resolution Microphones It’s been 25 years since the C800G, but Sony’s latest 50kHz-capturing microphones have been worth the wait. Review: Craig Field
NEED TO KNOW
The violin sounded so sweet through the Neumann U47 and custom-built Giles Audio valve preamp… However, we all chose the Sony
PRICE Expect to pay C100 - $2595 ECM100U - $1795 ECM100N - $1650
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CONTACT Dynamic Music: (02) 9939 1299 or info@dynamicmusic.com.au
PROS Impeccable detail Natural frequency extension Consistent performer
CONS None
SUMMARY It’s been a long time since Sony delved into high-end microphone design. The last time it did, it released an instant classic, the C800G. While these new mics are focused on bridging the gap with Hi-Res audio, they’re simply astoundingly transparent mics.
The sleeping giant has woken. Sony has finally released new studio microphones, its first new studio-specific designs in over two decades. So why now? Is it just having a corporate mid-life crisis, reminiscing over the golden days of mics like the C37? Or was Sony’s revered C800G thermoelectrically cooled tube microphone (released 25 years ago) just too hard of an act to follow? To understand the reasoning behind Sony’s new microphones, we perhaps need to dig deeper into the guiding principles of Sony’s entire audio vision. Unlike other manufacturers racing to the bottom. Sony Music doesn’t just support Hi-Res audio, it’s built an entire eco-system around it. It makes Hi-Res audio players, both for home and portable, it makes some very nice-looking signature series Hi-Res players and a range of Hi-Res audio headphones which have a frequency response from 4Hz-120kHz. There’s actually no universal definition for Hi-Res audio, though generally speaking it is uncompressed audio delivered in a higher bit rate and sampling rate than CD quality; 16-bit/44.1k. Typical Hi-Res file formats would be 24-bit/96k, 24-bit/192k or 1-bit DSD or DXD. It’s polarising, and I could open a can of worms. Briefly, there is a mathematical theory called the Nyquist Theorem. You probably already know it, but it says if the highest frequency component (say in Hertz) for any given analogue signal is fmax then to adequately reproduce that signal in the digital domain the sampling rate must be at least 2fmax. If your sample rate is 44.1kHz, like a CD, the maximum frequency component it could reproduce would be around 20kHz, which is the top of the human hearing range. Or a few kHz above for many. It’s why CDs use a sampling rate of 44.1kHz. On the bit depth side, it’s linked to dynamic range. Now, why would you need to go any higher than 16-bit/44.1k? Well, there’s a case to be made for having a longer slope to your anti-aliasing filter, which a higher sampling rate affords; there’s also intermodulation distortion; quantisation noise; and loads of other things you can obsess over once you crack the seal on this particular can. The takeaway is that a higher sample rate and higher bit rate yields a greater frequency range and larger dynamic range. Whether you can hear it is up to you. SONY’S NEW CLASS
Sony has a big stake in the Hi-Res audio market, manufacturing sound systems for listeners to enjoy Hi-Res recorded works means it supports Hi-Res content creation by labels. That support led it to recognise a possible weak link in this high-end audio chain. It’s no longer file formats or playback systems, it’s the capturing device itself. While interfaces, DAWs, plug-ins and even some digital monitoring systems have embraced this rarified air, microphones have rarely bothered to focus on anything humans can’t hear. Sony wanted to do something about that, and has created three new Hi-Res-capable transformer-less microphones: the C100, the
ECM100U and the ECM100N. I had the opportunity to try these microphones out in my Blue Mountains studio, where the C100 arrived in a very sturdy plastic protective case accompanied by a purpose-built shock mount. At first glance it looks like a standard side-address large-diaphragm condenser, but inside holds a rather unique snowman-like dual-capsule design. The 25mm bottom diaphragm is designed to capture a response range of 20Hz to 25kHz, while the smaller 17mm capsule on top captures 25kHz to 50kHz. The microphone is very well built and incorporates some of the anti-vibrational design of the C800G microphone. It’s not a large microphone at all, which is a great advantage in studio work. The C100 has three available polar patterns: omni, cardioid and figure eight, accessed by a firm switch mounted just under the logo. The first instrument I recorded with the new C100 was an acoustic guitar. I recorded the guitar with the microphone in cardioid and pointed slightly towards the lower edge of the bridge. In cardioid the microphone is very focused. However, the guitar was being tracked solo so it didn’t afford a great deal of insight regarding off-axis or rear rejection. Using a Maselec preamp, the guitar sounded very clear and clean. The image was quite focused but never felt unreal or too forward at any point. I went on to record violin and uke with the C100. All the recordings were airy, open and true, without a definable mid presence or peak. My studio is very focused on piano recordings and we are fortunate enough to have one of the finest concert grand pianos in Australia, in our main room. It’s always a good test for any equipment, be it microphones, preamps or monitors. It was here that I really put all three new microphones to the test. TRIO OF TESTS
I had a trio of some of Australia’s very finest players coming in for nine days of recording. We were recording new modernist compositions from Australian/New Zealand composer Alan Griffiths with concert pianist Nicholas Young, violin played by Dominik Przywara, and orchestral musician George Yang on cello. Sony didn’t have enough production models available to send a complete kit to me at the time of recording so I miked the concert grand with my own studio microphones. I used the C100 alongside a Brauner VM1 on the cello, both in cardioid as it was in the same room as the piano. The violin was isolated and mic’d with both Sony ECM 100U (cardioid) and ECM 100N (omni) smaller condenser mics, as well as a vintage 1974 Neumann U47. One of the challenges I most enjoy when recording instrumentalists is they intimately know the colour and nuances of their instruments and how they should sound. Once we were finished tracking, we all sat together to listen to the new microphones. The Brauner VM1 is a fantastic microphone. It’s a modern design that’s also valve. I don’t think we can sincerely compare apples to apples, but after
MIC POSITIONS For a look at Craig’s hybrid Decca Tree setup and Blumlein positioning, check out this video filmed during the sessions. youtu.be/_WyqpE6PK88 FUTURE IMPRESSIONS
Is High-Res the future of music? I’m not the person to answer that, though I sense many of us who have been working in this industry for years can sense we are still a little in the wilderness. What is apparent, though, is a huge sense of nostalgia in the recording industry, perpetuated by re-releases of classic equipment, both in hardware and software emulations. This might reveal a longing for a time when we knew where we were heading, or perhaps a lack of imagination. Either way, Sony has made three modern designs for the modern era of sound recording and whether you’re sold on Hi-Res audio or not, they’re still great mics. We should all be thankful for a new sound, a new song, a new day.
a few listens all of us chose the C100 in cardioid over the Brauner. The Brauner is superb but the C100 actually seemed more open, with more space around the instrument. The cellist, George, played with great passion and strength, and the C100 held the accuracy and SPL of this ferociously played cello, whereas at times, the Brauner seemed to be ever so slightly gathered in the 1.5kHz region. This could likely be corrected with a little more distance from the source. We could have chosen either recording but used the Sony for the final mix. The violin sounded so sweet through the Neumann U47 and custom-built Giles Audio valve preamp. The mic is in mint condition, so rich and musical. You’d be hard pressed to find a geezer who wouldn’t think that combo of virtuouso violinplaying and vintage mic sounded great. However, we all chose the Sony ECM 100N omni. Its detail, ability to handle the extreme highs of the violin, and overall accuracy in capturing the dynamic range and intensity of the playing was outstanding. The cardioid we had in the room sounded good, but was possibly not in the correct position and distance from the violin for it to be in the running. Again, we could have used the genuinely beautiful Neumann recordings, however, all our ears immediately gravitated towards the Sony. THIS TIME IN STEREO
The following week I was back in the studio recording a classical pianist and violinist. By then I’d received an additional C100 to try a Blumlein setup on the piano, which is ideal for that genre of music. On the violin was an augmented Decca Tree setup I have used a number of times in my studio for classical violin recordings. I had a pair of the ECM 100N omnis in a close stereo spaced AT 43
The cardioid version (ECM 100U) of Sony's hi-res pencil condenser is sensitive all the way out to 50kHz.
NEUMANN VS SONY Mastering engineer and tech guru Dave Askew helped fill out the picture of what the Sony mics are actually capturing by comparing Craig’s piano recordings using a stereo pair of Neumann KM84s versus a pair of Sony C100s . As you can see, there’s barely anything going on above 20kHz or so on the Neumanns, yet there’s a remarkable extension to the Sony sound. For a better look at Dave’s process, let him walk you through his comparison in a video at youtu.be/wzDAijS3nJc
pair with the ECM 100U (cardioid) as the centre forward-projecting microphone. The way I use this array is still in the equilateral triangular shape, but I move the inner cardioid microphone to focus on capturing some of the atmosphere from directly above the player. It’s slightly off axis and not pointing perfectly towards the violin, which takes some work to get right. Too much here or there and the image is out and the violin moves from left to right. Get it right and it sounds extremely natural and the violin seems to inherit a space rather than be the centre of an image. The C100s performed exceptionally well, netting a little wider image than previous microphones I’ve recorded with. I really took my time with the setup and felt I had the right length of note for the space and music being recorded. The definition in the lower bass notes from the piano was exactly what you could hope for when recording an instrument like this. The microphones easily handled the huge range and sound pressure of the concert grand; you could not hope for a more transparent and lovely image. My homemade mongrel Decca Tree worked well too. The violin can at times be very hard to capture as a sound recordist. It can also be quite a subjective sound; it’s a good idea to listen well and AT 44
ask the violinist about their preferred elements of their instrument. Last year, my studio was involved in a very large project where we had almost 24 weeks of continuous violin recordings and mixing/ mastering. Needless to say, I have some experience with the instrument. Another joy of this array is that when placing an accompanying instrument, you can very subtly place the image slightly to the left or right depending on how it naturally sits to your ear. This gives a very realistic and true positioning of the instruments, even though they are recorded in isolation, in two separate rooms, with two different atmospheres. The two smaller Sony microphones utilise the 17mm diaphragm from the C100 design, and work in the range of 20Hz to 50kHz. There would appear to be no real points in that frequency range where anything is particularly exaggerated, though the cardioid is obviously quite different to the omni. These are both sensational microphones. The challenge of capturing the very upper register of the violin was effortlessly overcome with both designs and the silkiness of those higher frequencies was a real joy. To sum up, I must say that recording with all
three members of Sony’s new High-Res series of microphones was a great success. They seem to have limitless openness when capturing sound sources and I doubt any instrument would be too hard for them. The C100, I believe, is a very interesting design and a studio all-rounder. I did not get to try it on vocals but the way it captured the stereo image of our studio’s concert grand was, in a word, stunning. FUTURE IMPRESSIONS
Is High-Res the future of music? I’m not the person to answer that, though I sense many of us who have been working in this industry for years can sense we are still a little in the wilderness. What is apparent, though, is a huge sense of nostalgia in the recording industry, perpetuated by re-releases of classic equipment, both in hardware and software emulations. This might reveal a longing for a time when we knew where we were heading, or perhaps a lack of imagination. Either way, Sony has made three modern designs for the modern era of sound recording and whether you’re sold on Hi-Res audio or not, they’re still great mics. We should all be thankful for a new sound, a new song, a new day.
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REVIEW
AVID PRO TOOLS | MTRX Audio Interface
When you need a hand — or a 1500 x 1500 Pro Tools matrix — call a DADMan. Review: Mark Davie
NEED TO KNOW
‘DADMan?’ Was all I could think about when I installed the control software for the Pro Tools | MTRX interface from Avid. I mean, it’s hilarious, completely fuddy duddy, but what’s up with that? Avid is nothing if not particular about its software releases, and DADMan sounds like an in-joke or development codename that slipped through to the keeper. Well, as it turns out, while the MTRX faceplate might be branded Avid in glowing purple splendour, inside it’s all someone else’s work. The MTRX is essentially a Digital Audio Denmark (DAD) AX32 in Avid guise, with Digilink connectors pre-installed on the rear. The AX32 is a first choice interface for post houses, often spec’d alongside large consoles as a means of whizzing around scads of digital audio channels as well as having Digital Audio Denmark’s best AD/DA conversion for the few analogue channels they might need — monitoring and the occasional mic input. Previously, the AX32 carried Digilink ports to
CONTACT Avid: 1300 734 454 or steve.chavez@avid.com
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interface with Pro Tools HD. Those have since been nixed from the AX32. Now, you simply buy the Avid version, MTRX, with Digilink pre-installed. TAKE ANY ROUTE
While hilariously named, the DADMan is one serious piece of routing software. The program lets you wrangle a possible matrix of 1500 x 1500 I/O at sample rates up to 192kHz. It’s all designed to be controlled via ethernet, and it has redundant internal power supplies, so MTRX can sit in a server room on a network somewhere and serve as the the hub for all your I/O. To get you started, there’s a whole range of onboard digital I/O that comes built-in to the chassis. There are eight pairs of AES, 64 channels of MADI I/O on coaxial BNC connectors, and the aforementioned dual DigiLink Mini ports, for
PROS Impeccable PCM & DSD conversion Neutral high-gain preamps Loads of digital I/O & option cards Monitoring options up the wazoo
CONS Outdated DADMan GUI
a total of 64 I/O or routing 32 I/O to two separate rigs. You can sync via worldclock I/O over BNC connectors, which also carries a black burst for accurate video sync, or use the AES11 input. If that’s not enough I/O, you can also install up to eight option cards, which can either be eight line inputs, eight line outputs, a combination card that allows you to have eight mic or line inputs, or a two-channel version of the same combo card. On the digital side, there are options for eight AES3 I/O, 64-channel Dante module, dual MADI I/O if you need to keep racking up the channels quickly, and a dual SDI/HD/3G card for up to 1080p/60 video input. On the chassis, you can also add more MADI via optical or Embrionix’s HD-BNC-type hot-pluggable modules.
SUMMARY Avid’s MTRX is the link between every type of connection, Pro Tools, and Avid’s line of mixing surfaces. If you’ve ever felt limited by what you can get in and out of ’Tools, then MTRX is the answer — MADI, AES, Dante, and all controllable over Ethernet. The Digital Audio Denmark conversion just ensures all your I/O is the best it can be.
INSIDE DADMAN
I’ve been using the MTRX in replacement of my HD I/O setup for a couple of months now, and while I don’t fit the typical post-production profile of a MTRX or AX32 user, it’s been a simple transition despite having to do things outside the Pro Tools GUI. With an active ethernet connection, it took a moment for the DADMan to appear. Once discovered, it never failed to appear in subsequent boots. The GUI isn’t particularly lovely. It’s a fairly rudimentary bitmap that doesn’t scale, and looks poxy on modern screens, but it’s very efficient. While there was no marking to indicate whether my eight-channel analogue input card was the mic/ line version, or simply line, it was immediately obvious once I opened DADman. Big fluoro buttons switch between mic and line. It flipped the interface between a long gain fader stretching from -18dB to +72dB when functioning as a mic preamp, and a series of dBU values in 3dB steps for the line input. These selectable input levels are further definable by a ±1.5dB slider that travels in 0.1dB increments. You can also send +48V phantom to the mic preamps, and phase flip or mute both mic and line. There’s also some pretty solid peak metering. On the output side, the range extends from -60 to +24dB in 0.1dB steps. There’s also a shortcut for snapping to +18dB by holding the Command key and clicking on the fader. As the name suggest, the matrix is where the action happens. You simply click on the squares that intersect an input and output destination, which expand to show the available I/O to route. Everything is clearly labelled, and the process is dead simple. The other heavily configurable area is the Monitor Profile box. You can route multiple input and output sources into different monitoring setups, all the way up to 64 speakers for Dolby Atmos. Route monitor sends for talent that you can shoot back over Dante. You can also setup folddown relationships, like sending a 7.1 to stereo. You can separately dictate whether each channel goes to left and/or right, and set how many dB down it should be from the original level. The user configurable monitor section is controllable via Eucon, from the Pro Tools control iPad app I was using, all the way up to the flagship S6 console, where it integrates directly into the monitor control section.
The DADMan software packaged with MTRX lets you easily route within a 1500 x 1500 matrix.
You can see why it’s a favourite of the postproduction industry, but it can also solve a lot of problems for the studio owner, especially with the simple Dante routing, or anyone dealing with high channel counts, or multiple monitor configurations. COAST IN NEUTRAL
From a sonic point of view, the conversion on this box is unquestionably great. Digital Audio Denmark is renowned for developing crystal clear conversion. Despite Pro Tools’ sampling rate limit of 192k, the MTRX is capable of 384kHz in PCM, it also does DSD-64 and DSD-128 (1-bit sampling at 64 or 128 times the frequency of a standard CD). Perhaps Pro Tools’ future might include these higher resolutions. When comparing the preamps with other highend interface preamps, the MTRX’s gain range is out of reach of most. However, the interface preamps do what most good ones seem to do these days; it records transparently, without flattering. I used a Shure SM7B to record some vocals, a mic that many interfaces struggle to record noise-free with their limited gain ranges. No such issues with the MTRX. Switching between the MTRX preamps, a Sebatron tube pre, and a JLM Audio 990 op amp-based preamp, the MTRX had plenty of low end, but sounded less impressive than the forward, harmonically-rich recording of the Sebatron. The JLM delivered a similarly throaty rendition, but sounded a little less pinched than the Sebatron in the low end. The MTRX remained neutral, refraining from adding any specific characteristic to the recording. I didn’t miss that character when recording drums. With so many different pieces of the instrument to pick up, a neutral pre can be beneficial. It doesn’t favour the low end of the kick over the crack or punch of the snare, or vice versa. It just worked; the big room recording sounded
deep and punchy, and fit perfectly into a folk mix with just a little Eventide Omnipressor bus compression. We recorded a few passages alongside Focusrite’s top of the line Red 8Pre interface, and overall we found the MTRX to have an ever so slightly more detailed top end. However, similar to our tests between the Apogee Symphony and Avid HD I/O conversion, it was a matter of taste — perception of warmth versus infinitesimal details. CONNECT ’EM UP
The MTRX is going to be a no-brainer for some; it’s the best way to interface post workflows with Pro Tools, and a great way to onboard MADI and Dante into your signal chain. For the studio crowd, it’s a much more flexible device than a standard HD I/O, given the user can freely switch between mic or line input on the analogue input cards, whereas you can only add line inputs to a HD I/O. Its flexible routing, expansion options and monitoring control makes it a formidable box for a 2U device. It simply does everything, at the best quality possible.
PRICE Pro Tools | MTRX Base unit: $6,208.60 Option Cards: 8 Line (AD or DA): $2,896.61 2 Mic/Line AD: $1,759.49 8 Mic/Line AD: $4,138.61 8 AES3 I/O: $2,275.61 Dual SDI/HD/3G: $3,793.61 64-channel Dante: $793.50 Dual MADI I/O w/o SFP: $2,896.61 MADI Module for Base Unit: $454.02
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REVIEW
PMC RESULT.6
Nearfield Monitors PMC has finally developed a more affordable studio monitor so the rest of us can get in the transmission line.
NEED TO KNOW
Review: Brad Watts
PRICE $4499/pair CONTACT PMC Australia: (03) 9426 3660 or pro@pmcspeakers.com.au
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PROS Superlative bottom-end Great imaging and deep soundscape Simple no-frills operation More affordable
CONS Possibly a little large for some
SUMMARY Transmission line speakers have largely been exclusive to the top-tier. PMC designed a new one from scratch to bring its ATL-style of speaker to the masses. The result.6 still has all the trappings of a PMC speaker — accurate, efficient low end; a wide sweet spot; and faithful reproduction — just without the DSP trappings.
It’s not often a set of PMC monitors make their way onto my monitor bridge. I believe the last PMC monitors I auditioned were Digidesign RM-1s — a collaborative effort between the two companies from way back when. Regardless, each time I do sit in front of a pair of PMCs, I need to re-calibrate my perception of lower frequency reproduction. Why is that? Well PMC does things a little differently to your usual monitor manufacturer. All PMC’s designs utilise what’s known as a transmission line as part of the monitor cabinet design. The upshot of incorporating a transmission line within the cabinet is the taming of reverberation caused by the low-end driver interacting with the rear of the cabinet — as you’d find with typical bass reflex designs, ported or otherwise. Consequently, the monitor projects bottom end in a truer fashion. With a non-ported cabinet the transmission line will dampen and absorb unwanted lower frequency anomalies, and with a ported transmission line those lower frequencies can be altered to leave the open port in phase with frequencies emanating from the front of the low-end driver. The transmission line can be any shape, and is often ‘folded’ to increase the length of the line to a suitable distance, and to keep it within the confines of a usable cabinet size. The effect is an extremely tight, uncoloured, and balanced low-end, which as I mentioned, takes a little getting used to. Initially the effect is almost as if there’s low-end actually missing, but once you settle in to the PMC sound you realise everything, and more, is there as it should be — it’s just more faithfully reproduced. Such has been my experience. LINE ’EM UP
Transmission line cabinets aren’t new. The concept was pioneered by Benjamin Olney during the early 1930s while working for Stromberg-Carlson Telephone Co. This design was dubbed as an ‘acoustical labyrinth’. Contemporary designs evolved during the mid 1960s with research and design by Dr A.R. Bailey and A.H. Radford. While these designs were initially based on the acoustical labyrinth work, it was Bailey who hit upon the idea of using dampening to further reduce resonances along the transmission line. From there the history gets a little murky, with American, Irving M. ‘Bud’ Fried and a British group comprising John Hayes, John Wright, and David Brown designing and marketing transmission line designs primarily under the IMF brand (Irving M. Fried’s initials). Unfortunately the IMF company name became a point of conjecture, with the British developing IMF transmission line cabinets and speakers, while Bud kicked about the U.S. distributing the Britishdesigned speakers along with selling lesser quality designs. Once the legal dust settled, Bud created the Fried company in the States, while the three Brits entered a joint venture with driver manufacturer Elac, naming the company TDL. Into the mid 1970s, a chap by the name of L. Bradbury commenced a spate of work researching the effect of different absorption materials for
A cutaway of the transmission line design at the heart of the Result.6’s extended and efficient low end response.
lining transmission line walls. Unfortunately he kept hitting a wall, with his dampening materials, primarily wool and glass fibre, resulting in quite variable measurements. Climate, humidity, and changes to the material over time rendered many unrepeatable characteristics. It turns out the choice of dampening materials have as much to do with tuning the cabinet and transmission line as the cabinet itself, consequently transmission line cabinet design becomes a bit of a dark art. Understandably there’s very few manufacturers keen to sink research and development resources into such a finely tuned system. At any point in the history of transmission line speakers there’s only a handful of designers prepared to tackle the concept. For those that are the outcomes are impressive, which is why you’ll find it’s primarily esoteric audiophile manufacturers developing transmission line speakers. The standout player in this tiny field is PMC. Not only does the British firm aim designs to the audiophile market, with its deeper pockets and disposable income, its grass roots are firmly footed in the studio and recording markets. Since 1991, PMC founders Peter Thomas and Adrian Loader have dragged transmission line design firmly into the 21st century, and have dubbed their designs ‘ATL’ or advanced transmission line. Peter’s BBC training led him to create speakers initially for the BBC and London’s Metropolis Studios. Those first speakers are still in use today at the BBC and Metropolis, and now thousands of other top-shelf studios worldwide.
HEAR THEM AT YOUR PLACE As you can tell from Brad’s review, if you’re not experienced with transmission lines, it can take a little time to adjust. To help you make a decision, PMC Australia has demo pairs of its range available to AudioTechnology readers. If you want to try them out in your own space, just give them a buzz or email on: (03) 9426 3660 or pro@pmcspeakers.com.au
TRICKLING DOWN THE LINE
Knowing the history and some of the design fundamentals lets you realise transmission line designs aren’t cheap. It’s why it’s taken 27 years for PMC to develop an affordable monitor design it could stick its badge on — the result6. Like all PMCs the result6 is a transmission line design, or ATL as PMC do, so it’s a little deeper than your average bass-reflex cabinet. But more on that later; let’s have a look through the specs and build. The result6 is a two-way nearfield design incorporating a 6.5-inch doped fibre low-end driver and a 27mm soft-dome high-frequency driver. Amplification is a fairly modest 100W for the low end and 65W at the top end. However, transmission line designs make the bass driver work more efficiently, requiring less power to deliver the same low end as a bass-reflex design. AT 49
Both amplifiers are a Class-D design, which we know offers the advantages of low heat production along with smaller size and less weight. There’s little else in the way of published specifications for the amplification apart from the mention of the amplification compensating for driver impedance variations due to frequency changes. On the other hand, there seems to be more information regarding the upper level twotwo series monitor’s amplification, so I suspect this is an area that has helped PMC keep the final price down. Suffice to say, there’s no changing the amplification — it’s built in and does the job well. FIN BODY WORK
Looking around the largish cabinets you see a rubberised strip encircling the cabinet at both the front and rear. This provides a non-slip and de-coupled footing for the monitor and would function whether the result6 is sitting vertically or on its side, although PMC suggest vertical positioning is the correct procedure, as would I. Regardless, if horizontal placement is your thing, the rubberised bands will work just as well in this configuration. The entire cabinet is almost as deep as it is high, with measurements of 380mm high by 199mm wide, with the depth stretching out to 360mm. Larger than the PMC twotwo.5 monitor and closer in size to the twotwo.6. They take up quite a bit of space so this will be a consideration if you’re tight on control room real-estate. The depth is obviously exacerbated by the size of the transmission line architecture, which occupies a good 85% or more of the cabinet depth. If you decide to mount the result6 monitors on stands you’ll need to check the stands are up to the task. Otherwise there are dedicated PMC wall-mounts designed specifically for the result6, along with PMC’s high-mass tube104 stands that will keep the monitors stable. That said, the result6 is actually fairly light for its size at 8kg. Looking further into the driver and porting arrangements, the result6’s transmission line exits AT 50
the cabinet from the front baffle below the low frequency driver, which is covered with a fine mesh grille. I’m all for front porting in any monitor, but it’s especially important with a transmission line as the line behaves almost as a low-end driver itself. It’s also advantageous if you’re keen on soffit mounting, PMC suggests the result6 to be a perfect candidate for surround monitoring systems, which is where a soffit mount may come into play. At the high-end is the previously mentioned 27mm soft-dome tweeter. And as soft-dome tweeters should be, it’s not fatiguing, and my personal preference (give me ribbons or softdomes — I can’t do metal tweeters). All seems relatively standard fare until you notice the radial fins either side of the tweeters. This is PMC’s latest weapon against diffraction effects cause by large amounts of front baffle surface area surrounding a small driver, and indeed, typical cabinet edge diffraction effects. Trademarked as ‘D-Fins’, they jumble high frequency radiation patterns, resulting in a wider sweet-spot and a less ‘smeared’ high-end reproduction. The D-Fins certainly seem to make a difference with a crisper and more precise image than I’d expect from a more typical soft-dome arrangement. And as the manual says, the imaging and sweet-spot is remarkably wide. I found the high-end incredibly accurate, whereas I was initially expecting that high-end wash you find with wave-guide style high frequency dispersion. It seems such a simple idea, but it works. IMPRESSIVE RESULTS
So why the lower price point, and where have the corners been cut? What’s missing? Again simplicity is the key. Unlike many monitors released over the last few years there’s no DSP. There’s no digital I/O, no tone-shaping, no corner EQ settings, no bass-lift, no room-adaptive spatial enhancement malarkey, and no fuss. If you need that stuff you’ll need to pony up for the twotwo.6 monitors. To be honest I feel the recent array of monitors touting umpteen different methods of balancing a monitor via DSP goes against the grain. It reminds me of
the old ’70s idea of running a 31-band graphic EQ prior to the monitor sends to supposedly tweak the speakers to suit the room. If you look out the back of the result6 there’s a power input and switch, an XLR input, and an attenuation pot — continuous — not stepped, which I also like very much. Many a time I’ve wanted finer adjustment in monitor attenuation and have had to pull in ancillary hardware, adding additional connectors and normally unnecessary signal path. What hasn’t changed? Well, being PMC monitors they are ‘Designed, made and tested in the UK’. I had to double check the result6 wasn’t emblazoned with something along the lines of ‘Designed in the UK and made in China’. But no, PMC are true to the cause, and it speaks volumes for the company’s commitment to its craft. Hats off! The results are impressive, and yeah, that pun has to remain in this case. Like PMC’s other ATL monitors, the bottom end is controlled, tight, realistic and distortion free, and stable throughout the SPL range. Like I said, it takes a little getting used to at first, but then that’s probably because you, like me, have been listening to bass-reflex designs for too long. PMC has far too many runs on the board to be taken lightly. The company’s website will guide you through how many household names are reliant on PMC monitoring (Studios 301 included) along with many that aren’t — the back-room names behind the names, the mastering rooms, the studios. The result6 is about to make that list a whole lot longer.
AT 51
REVIEW
UNIVERSAL AUDIO ARROW Thunderbolt 3 Interface It’s now easier to get yourself onto the UAD Powered Plug-in platform. Review: Christopher Holder
Arrow is a beautiful piece of industrial design. The gun metal grey, steel chassis, the metering, the big knob; it’s all solid. It’s a desirable looking piece of kit. Think of it as a primo $700 desktop interface with great 24-bit/192k converters, and two topdraw preamps, and Arrow is already shaping up as a high-ranking wishlist item. But we all know that’s not really where our desires truly lie. It’s the UAD plug-ins. These are the plug-ins that have ‘turned’ rusted-on, analogue die-hards; plugs that are defining the sound of top-quality commercial productions the world over. SHARPEN UP
NEED TO KNOW
Arrow promises 11 UAD plug-ins straight out of the box, including a Marshall Plexi Classic Guitar
PRICE $699 CONTACT CMI (03) 9315 2244 sales@cmi.com.au www.cmi.com.au
AT 52
PROS UA preamps Excellent converters 11 top-shelf plug-ins (worth over $1000)
amp emulation, LA-2A and 1176 compression and Pultec EQ. It’s a super-tasty smorgasbord and well worth Arrow’s price of admission. The plugs are selected for Arrow’s buspowered ‘record anywhere’ ethos. Namely, there’s loads of scope for applying these plug-ins during an ad hoc tracking session, not just as insert effects for the mix. UA is marketing Arrow as the world’s first Thunderbolt 3 interface for Mac and Windows. This works fine for me as I’ve recently purchased a mid-spec’ed MacBook Pro that’s dispensed with boring real-world I/O (ethernet, USB etc) in preference to a parsimonious complement of two USB-C holes (gotta love Apple). Getting the Arrow working was about as easy as connecting up the Thunderbolt 3 cable (not included, and still new enough to be
CONS Thunderbolt 3 cable not included
stupidly expensive), registering the device and downloading the installer/plugs. Note: you can’t use any ol’ USB-C cable, it’s got to have the Thunderbolt logo on the connector. THUNDER STRUCK
If you didn’t get the memo: Thunderbolt 3 uses the USB-C plug, but delivers lightning quick 40Gb/s data transfer (USB 3.1 operates at 10Gb/s, by way of comparison). Thunderbolt 3 is bi-directional with four lanes of PCIe Gen3. It appears to be a transmission protocol well suited to the world of audio. Arrow isn’t a device that’s going to push the bleeding edge of Thunderbolt 3’s limits. It’s a 2x4
SUMMARY A true, go-anywhere recording interface using the lightning-quick Thunderbolt 3 protocol. What’s more, Arrow allows you to join the UAD Powered Plug-in family for under a grand.
F1 FIELD RECORDER + LAVALIER MIC PROFESSIONAL AUDIO. UP CLOSE.
The Zoom F1 Field Recorder + Lavalier Mic is the perfect solution for videographers who need to capture the nuance of every word with professional quality audio. Superior Sound: Newly engineered LMF-1 lavalier mic+ (up to) 24-bit/96kHz recording to micro SD card. Low Profile: Attaches neatly on belts, waistbands, or slipped concealed into a pocket. One-Touch Controls & Display: Instant access to record levels, limiter control, lo-cut filter and volume output + sunlight viewable LCD. Videographer Essentials: Record Hold function, onboard limiter, battery powered (or power adapter), transfer files via card reader or directly via USB. interface. We may not be talking about recording a multimiked orchestra but the two Unison mic preamps combined with UA conversion and the real-time UAD plug-ins means you can monitor, for example, a hi-z guitar input through the Marshall guitar amp simulator at near-zero latency. It’s a delicious prospect for a bus-powered portable setup. I pulled Preshan away from his UA Ox review to monitor his Strat through the Marshall Plexi Classic amp sim, and a RealVerb plate. I also ran him through the Pultec EQ and the Raw stomp box emulation. The SHARC DSP chip was only ticking over and Preshan reported great ‘zero’ latency playability. I had a blast toggling through the various presets on the UA plugs. Every patch sounded amazing in its own way. There were no bad sounds. These are very classy plug-ins. Of course, you can call up your selection of UAD plugs from within your DAW. And why wouldn’t you? They’re likely to be better than anything you currently have in your arsenal. POWERED HUNGRY
By way of a minor niggle. The installer includes all 90+ UAD plug-ins; leaving it up to you to filter the ones you’re authorised to use. Otherwise a whole host of ‘zombie’ plugs that are off limits (unless you pull the credit card out) will sit there taunting you every time you want to insert an effect. My guess is Universal Audio sees a lot of authorisation sales after midnight and half a dozen beers, when acquiring that Distressor becomes a matter of life ’n’ death! STRAIGHT AS
Arrow is significant for a number of reasons. Firstly it’s the most cost-effective way to join the UAD Powered Plug-in family, and you’ll never look back. If you upgrade to an Apollo down the track your plug-in investment will follow you. Secondly, Arrow is significant for the fact it’s using Thunderbolt 3 for data and power — it’s nearly half the price of UA’s Apollo Twin and the Arrow doesn’t need a power supply. I can’t think of a more desirable and more capable interface for under a grand.
Proudly distributed in Australia by Dynamic Music
dynamicmusic.com.au
AT 53
REVIEW
KORG PROLOGUE 16
Polyphonic Analogue Synthesizer This 16 voice analogue synth has more oscillators than you have fingers — goodbye voice-stealing anxiety! It also has some hardcore digital tricks up its sleeve. Review: Jason Hearn HOT WHEELS — Korg has provided proper pitch and modulation wheels (unlike Minilogue’s modulation-stick), and their location above the key-bed facilitates a compact horizontal footprint on the desktop.
NEED TO KNOW
TOUCH & GO — The ‘Natural Touch’ velocitysensitive keyboard feels fantastic to play with - a light, fast action and minimal lateral wiggle, but it lacks aftertouch.
PRICE Expect to pay Prologue 8: $1999 Prologue 16: $2899
AT 54
CONTACT CMI Music & Audio: (03) 9315 2244 or info@cmi.com.au
SCOPE IT OUT — All parameter knobs are solidly chassis-mounted and for parameters warranting it, high resolution adjustment and recall is available. Like the Minilogue, the main OLED display has a dual-purpose as a real-time oscilloscope. It’s a real joy to see the effects on the oscilloscope while designing sounds.
PRO STOCK — There’s nothing to suggest Prologue is built to a cost and it feels quite heavy (10kg) considering its size; which isn’t surprising with the voice count on offer. The real wooden cheeks give a hint of vintage flair and the unassuming front panel doesn’t demand your attention, but rather, invites it.
PROS 16-voice polyphony! Goodbye voice-stealing Incredible sound Superb build quality & look Open API for usercontrolled oscillators & FX
CONS No modulation matrix Limited modulation possibilities Only two-parameter settings for FX No aftertouch
SUMMARY Not satisfied with putting a whopping 16 voices in an analogue poly synth, Korg added a fully-programmable digital Multi-Engine oscillator inside Prologue… emphasis on programming. It might lack a modulation matrix, but this synth will go from dreamy to snarly on the stage or in the studio, all without stealing a single voice from your polyphony.
VCF — The VCF is a two-pole (12dB/octave) low pass filter that makes use of Operational Transconductance Amplifiers (OTA). It’s capable of self oscillation and can be modulated in a bipolar manner from the EG. It has a rich, powerful sound with reasonably consistent output level when swept across the audible frequency spectrum. A drive stage provides two preset amounts of drive: 50% and 100%, with finer adjustment available by backing off the levels in the oscillator mixer. A fixed 500Hz low cut filter can tame the lowermids of your patches, and VCF keyboard tracking can be engaged at 50% or 100%. With resonance at self oscillation, the key tracking is scaled to be in tune (great for high resonance, white-noise leads).
It would be easy to characterise the Prologues as next-level iterations of the affordable four-voice Minilogue, but they’re considerably more advanced. Most notably, the oscillator section now includes a digital Multi-Engine oscillator that vastly expands the tonal palette and ability to create movement in the base tone of a voice. Prologue comes in two versions: Prologue 8 features eight voices with a four-octave keyboard; Prologue 16 is the ‘deluxe’ model and provides eight additional voices, a five-octave keyboard, LF compressor and expanded front panel controls. I was sent the Prologue 16 to pore over, which gave me a good look at everything both new synths have to offer. With all those voices comes a huge number of patch memories. There are 500 available, and the first 250 are occupied by factory presets that can be overwritten. It’s a lot of sounds, but Korg has devised a number of ways to filter the patches, including by Category, Alphabet, Like, Frequent, Envelope, Random, or one of four different Live Sets. Being able to ‘like’ presets while surfing the factory patches for favourites proved very convenient. The Live Set mode lets you use the program buttons during live performance to rapidly recall four banks of eight patches.
ADSR — The first ADSR envelope is dedicated to the VCA; the second assignable to the VCF and pitch of the VCOs. The decay and release times are generous (around 17 seconds for both), I found the attack time (approximately four seconds) would benefit from being longer for really epic rising pads.
FIND YOUR VOICE
Four voice modes control how notes are allocated to the voices with additional control provided by Voice Depth. Poly mode allocates a single voice to each note, and increasing Voice Depth slightly spreads dual voices to each note, with increased level and detuning. Mono mode monophonically allocates a single voice to each note. By increasing Voice Depth, a second voice is engaged as a sub oscillator with larger settings transposing the sub voice up to two octaves lower. Unison mode allocates all voices to each incoming note, stacked, with detuning increased by Voice Depth. Chord mode allocates voices to one of 14 preset chord shapes selected by the Voice Depth control. A variable Voice Spread function adds ear candy by spreading the voice allocations across the stereo field with increasing strength. In the case of the Prologue 16, having a 16-voice unison stack sounds truly massive and those on the quest for the ultimate super-saw sound will not be disappointed! DIY OSCILLATORS
There are two types of oscillators onboard; analogue and digital. VCO1 and 2 are thoroughbred analogue oscillators, each having
LFO — A single LFO provides three waveshapes routable to either oscillator pitch, oscillator shape and filter cutoff. Although lacking sample-and-hold, it’s capable of audio-rate speeds. The BPM rate setting enables external tempo synchronisation of the LFO with range adjustable from four bars through to 1/64th, but is missing triplet and dotted divisions. The LFO can be set to free running or Key Sync via a menu setting, however an LFO delay function isn’t present.
three waveshapes (saw, triangle, and pulse). The shape control provides wavefolding for the saw and triangle waves, and duty cycle adjustment (with a distinct zero-through point) for the pulse wave, it can then be modulated by the LFO. The pitch of VCO1 and VCO2 can be modulated by the second envelope either simultaneously or independently. Versatile modulation functions between these oscillators include phase syncing of VCO2 to VCO1 — yielding classic hard sync sounds — or ring modulation. VCO1/2 also has a variable cross modulation circuit. Engaging sync and cross modulation simultaneously creates some truly searing leads and aggressive basses. The third oscillator is a digital Multi-Engine oscillator, which stretches significantly beyond the VCO-based waveforms. It has three core modes. VPM Mode: 16 preset VPM/FM variants available; from simple bread-and-butter waveshapes (including proper sine waves) through to throaty, vocal-sounding options popular in modern EDM genres. Aside from the shape, six additional parameters are tweakable, including a self-contained AR envelope. User Mode: 16 user slots are open for developers to design custom oscillators in the near future. From the factory, the first slot is occupied by AT 55
‘Waves’: a dual-oscillator wavetable synth with a ring modulate-able sub oscillator and bit-crushing functionality. Waves’ source code will be published as an example to follow. Development is intended only for experienced programmers (requiring skill in DSP math and programming C++), however, Korg may provide means for dabblers to also experiment. Further details regarding the SDK and development tools are available on Korg’s site (korg. com/au/products/synthesizers/prologue/sdk.php). Noise Mode: The shape control brings tonal variation to the four noise sources, but the envelopes can’t modulate the shape, only the LFO can. Unlike the User and VPM/FM modes no further settings are offered in the menus. Furthermore, there is no means to do Sample and Hold modulation of parameters using noise as a source. Not surprisingly, the Multi Engine is siloed from the analogue oscillators. It’s not possible to combine it with VCO-based sync, ring or cross modulation functions. While it offers a versatile range of tones, what’s currently on offer worked better in a supporting role to the analogue oscillators. Once open to developers, I’m sure the breadth of custom oscillators will invert that relationship. DOUBLE WHAMMY
A hybrid analogue and digital synth wasn’t enough for Korg, the Prologues are also bitimbral. At the expense of a halved voice-count, you can have two sounds active simultaneously — main and sub — which can be layered, split, or cross-faded across the keyboard. Usefully, the arpeggiator can be independently assigned to the main or sub oscillators. Six arpeggiator patterns are available, with four octave range options. Surprisingly, it lacks a note division setting and remains fixed at 16th notes, which might suffice for a band performer using tap-tempo, but is inconvenient when slaved to external MIDI devices. Oddly, the analogue sync input for Korg’s Volca series does offer either 16th or 8th note divisions. As far as clock goes, the source can be derived from USB, MIDI, or Internal. Usefully, it can follow Clock appearing at the MIDI input while connected to the DAW via USB — great if you’re running a dedicated external MIDI clock generator to remedy sloppy DAW clock. AT 56
The sound of the Prologue 16 is sensational; there’s nothing quite like proper VCOs; 16 pairs, at that! Having so many voices on tap almost eliminates voice-stealing anxiety when performing complex passages. Simultaneous VCO1/2 cross modulation and hard sync, in tandem with the filter’s drive stage, makes the Prologue capable of massive, angry tones. With the Voice Spread function, pads float in front of your speakers. With the layer function, incredibly complex tones are available with relative ease. Like most analogue poly synths, it can be a challenge to conjure really up-front bass tones in the way a dedicated mono synth can. When replicating a plucky bass patch on my Sequential Circuits Pro-One, the Prologue performed admirably, but didn’t quite have the same presence. EFFECTING
The Prologue features superb-sounding dual 24-bit FX processors, which can be disengaged for a pure analogue signal path. The first offers four modulation effects: Phaser, Flanger, Chorus, Ensemble. With the exception of the rate and depth, parameter control is limited to selecting from a wide range of factory presets. The second offers either Reverb or Delay, with a similar lack of control. Thankfully, each algorithm sounds excellent and the streamlined approach is effective; but I did crave the ability to tweak further. 16 slots for User modulation effects indicates the possibility for programmers to roll effects of their own design. A tantalising prospect — programming skills required. On the movement side, quite simply, the Prologues don’t have a modulation matrix. Modulation capabilities can be best summed up as follows: a pair of Envelope Generators (the first permanently routed to the VCA, the second routed to the VCF cutoff and pitches of VCO1/2), the highly routable Modulation Wheel and Expression Pedal, and the LFO (which can be routed to either oscillator pitch/waveshape or the filter’s cutoff frequency, but only fixed amounts). Depending on the algorithm selected, the Multi Engine can offer additional modulation, but other than the shape control, is independent of the rest of the synth. With the Minilogue, you could fake having a modulation matrix by taking advantage of its
motion-sequence recording function. Surprisingly, the motion-sequence feature didn’t make it into the Prologues. Nevertheless, exclusive to the Prologue 16 is a low frequency compressor strapped to the output bus with simple gain control accompanied by a backlit VU meter. With minimal gain settings, it gives subtle punch and colour, ideal for evening out comping patches and pads. With substantial gain applied, it gives edgy presence and character to synth basses whilst reigning in dynamics. PROLOGUE EPILOGUE
The Prologue is a clear winner for the liveperforming keyboard fraternity, delivering an excellent range of classic analogue sounds through to more exotic tones afforded by the Multi-Engine oscillator. It feels fantastic to play and, on a sizeable PA system, is an experience to hear. With its compact shortboard format, and solid build quality, it’s destined to become a road-warrior favourite. The addition of an expression pedal should overcome the need for aftertouch, and serve a more versatile function since its value doesn’t reset to zero with each key release. I’ve often stated that the hallmark of a good synth lies in its modulation matrix and available modulation sources, but perhaps I’m expecting too much. After all, the absence of a modulation matrix hasn’t affected my ongoing love affair with the Roland MKS-80. The Prologue 8 is pitched at a price comparable to Novation’s Peak, which isn’t a true analogue synth and lacks a keyboard, yet has a modulation matrix and more modulation sources. Still, the possibilities of Korg’s forthcoming Multi Engine oscillator and Modulation FX open-development can’t be found elsewhere. The larger Prologue’s 16 voices of analogue VCOs is a market first. If you’re a lover of pads dripping with analogue warmth and love playing big chords across both hands without concern for voice-stealing, the Prologue should satisfy in a manner which other analogue polys can’t touch. Its price will draw comparisons with DSI’s Rev 2, however, thanks to its high polyphony, it will stand proudly as The Player’s dream synth, destined to be a modern-day classic.
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AT 57
REVIEW
ARTURIA MINIBRUTE 2 & 2S Analogue Synthesizers The battle for monophonic synth supremacy just got a whole lot more brutal. Review: Preshan John
NEED TO KNOW
Few things cause as big a stir in the AT office as unboxing a new analogue synth. The unveiling of Arturia’s MiniBrute 2 and 2S was no exception. We’ve come to expect big things from this forward-thinking French company’s small packages. Needless to say the two synths were out of their boxes quickly. You instantly notice the larger size of the MiniBrute 2 thanks to the sprawling panel of knobs, faders and switches and luscious wooden endcheeks, which are basically a given on Arturia synths now. Both synths are solidly built with a metal chassis and rubberised knobs. The MiniBrute 2 has a nice keybed with a fast action while the 2S has a row of 16 backlit pads designed for a sequencer-based workflow. Both Brutes have a patch matrix in the top right corner and Arturia includes a helping of cute fabric-lined patch cables in the box.
PRICE Expect to pay $999 CONTACT CMI Music & Audio: (03) 9315 2244 or www.cmi.com.au
AT 58
PROS Powerful analogue oscillator-shaping Patch matrix adds tone & integrations Deep sequencing flexibility on the 2S
2 VS 2S
The top halves of both MiniBrute 2 and 2S are identical. Of the two oscillators, the first offers the most flexibility. A combination of sawtooth, square, and triangle waves are at your disposal, along with a noise generator. Mix in each to taste using the first four faders. Each wave has a degree of modulation, meaning you can get creative
CONS None
right from the outset. The Ultrasaw Amt knob thickens the sawtooth wave, Pulse Width and PWM knobs lets you modulate the square wave, and the Metalizer knob adds complex harmonics, specifically to the triangle wave. The second oscillator is more basic; offering either a sine, sawtooth or square wave via a three-way switch. Alternatively, you can use it as
SUMMARY MiniBrute 2 ticks all the boxes for a monophonic analogue synth — thick tone, loads of modulation options, enhanced by a patch matrix, and an extremely flexible sequencer on the 2S. MiniBrute 2 will be friendly to the uninitiated and a pleasure for synth fans looking to expand modular setups.
It doesn’t matter if you’re setting up your first remote live stream session, DJing your hundredth party, or mastering your twelfth studio album—the Studioseries has got you covered. All of these interfaces boast as much I/O as we could
Studio 24: 2x2 USB-C audio interface
squeeze into their sturdy chassis. But robust hardware is only half the package. The included Studio One® Artist
Studio 26: 2x4 USB 2.0 audio interface
DAW lets you record and mix with unlimited tracks and effects in an intuitive UI that stays out of your way. And the Studio Magic Plug-in Suite expands your sonic palette with powerful VSTs from the best developers in the business.
Studio 68: 6x6 USB 2.0 audio interface
Meet the family.
Regardless of what your musical passion is or where you make it happen—we’ve got the right interface for both your workflow and workspace. Learn more at www.presonus.com.
STUDIO-SERIES RECORDING SYSTEMS
Studio 1810: 18x8 USB 2.0 audio interface
Studio 1824: 18x18 USB 2.0 audio interface
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RACKBRUTE: A HOME FOR EURORACK MODULES Arturia has capitalised on the rsurgence of modular synthesis and the Eurorack standard with its new RackBrute product — a sturdy dwelling place for your Eurorack modules with a width of 176HP. RackBrute has an aluminium chassis complete with Arturia’s signature wooden end-cheeks. It’s available in both 3U and 6U racks. The built-in
an LFO by flicking down the switch to the right. Glissando time is set with the Glide knob. Both oscillators have their own tune knobs, or you can use the Global Tune knob in the Amp section to affect both at once. MiniBrute 2 excels at bass tones that maintain their chunkiness right down to the lowest octave. Lead tones are equally punchy. In the best possible way, every knob — from Glide to PWM — has a somewhat unpredictable analogue character. The sound will get gnarly, break up, slide around, in ways digital emulations sometimes struggle to match. There’s no shortage of modulation options. The Steiner-Parker filters are smooth and can be tied to an LFO easily. The Brute Factor knob goes from pleasing saturation to harmonic destruction. Both LFOs have five available waveforms and can run freely or be synced to the sequence tempo. IN SEQUENCE
The ‘S’ in MiniBrute 2S is for the built-in sequencer section. Each of the 16 pads is partnered with an endless, detented encoder. Its function changes based on whether you press the Pitch, Gate, Velocity or Press button. To create a sequence, first hold the Last Step button then select your desired sequence length with the buttons next to it — 16, 32, 48 or 64 steps — or any number in between by touching the pad you’d like as the last step. Make sure the Record button is lit, then, with Pitch selected, use AT 60
power supply feeds the modules with +12V/12V/+5V. Arturia Link, featured on the RackBrute, is basically a fastening system with a large degree of rotating flexibility so the actual rack can be positioned firmly at any angle. Combine that with the Y-shaped frame that can stand forward-facing,
the knobs to dial in a sequence while the note is displayed on the alarm clock-style LCD readout. Activated pads light up red. Once a tune is taking shape, repeat the process with Gate and Velocity selected to enhance the sequence. Note that you can tie two adjacent notes together by turning the gate all the way to 100%. You can save a bucketload of sequences and seamlessly switch between them by pressing the Load button with a corresponding pad. While it’s all very absorbing, there’s no escaping the time you have to spend figuring out how to fly through the sequencing process. It’s smooth-sailing once you’ve internalised all the button combinations and navigation. The little patch matrix and the included cables vastly expand MiniBrute 2’s creative possibilities. It’s laid out logically enough without stifling a healthy level of unexpectedness. Take an oscillator and use it to modulate a filter, or patch an LFO directly into an oscillator. Bring in an external synth, or sync the sequencer to an external clock. If you bought the key-less 2S, the patch matrix also means you can easily connect a controller to play in lines or access a greater note range than the single octave afforded by the 2S pads. GIVE IT FRIENDS
There’s absolutely no doubt that knocking up sequences and twiddling knobs is incredibly fun on the MiniBrute 2S, but after you’ve created a monster line or two, you have to ask “What now?”
backward-facing, or upright, and you’re assured of perfect ergonomics. While laying flat, MiniBrute 2 and 2S will fasten to the RackBrute via their Link connectors, so you can perch Eurorack modules directly above it — perfect placement to get the MiniBrute talking to a few modular buddies in close quarters.
Being analogue and monophonic, the MiniBrute doesn’t quite make it easy for artists to link sequences into songs on the fly. In isolation, you’ll doubtless come to a point when you have a jam playing back on the MiniBrute 2 and begin looking for the next step. Like an introverted genius — MiniBrute needs friends to bring out its best. Teamed up with a drum machine like DrumBrute, or integrating it into your existing MIDI setup will make a killer combo on the live stage. Alternatively, grab Arturia’s new Rackbrute, load it with Eurorack modules, and after hooking up a few patch points you’ll have a serious modular synth in no time. If you’re new to the world of analogue synthesis, MiniBrute 2 is a great first choice. The one-to-one nature of knob to parameter is a handy way of getting your head around synthesis. For the already initiated, the ability to mix together multiple oscillators makes MiniBrute 2 far more capable than most monophonic synths. If your collection is starting to grow, the work Arturia has done to integrate the MiniBrute 2 into the revitalised world of modular synthesis makes it a regret-free addition to your setup.
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REGULARS
LAST WORD with
John Meyer, Part I
John and Helen Meyer just celebrated 50 years of love that began when the young inventor showed Helen what Sgt. Pepper’s sounded like on a great set of speakers. The two have been introducing the rest of the world to great speakers ever since. Recently, Meyer Sound’s top tier descended on Sydney to celebrate its new alliance with Audio Brands. We sat down with the Meyers to talk about 12-year olds on radio, the Grateful Dead’s Wall of Sound and chasing audio holograms. Here is the first instalment.
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I grew up in an FM radio station in Berkeley called KPFA. I first started going there when I was seven. It was an educational station, and they’d do live theatrical broadcasts of Tom Sawyer and Huckleberry Finn. I could memorise the lines, so I got some bit parts. At 11, I got my own program telling fairy tales. I had speech coaches teach me how to tell a story on radio, create emotion and suspense. I’d do one for 15 minutes and use the spare 15 minutes to interview kids around the station. Kids of famous people in Berkeley like (musician) Larry Livermore and (film critic) Pauline Kael. Sometimes they wouldn’t show up, so I wanted to have a second fairy tale in the can. I learned how to remote control the tape recorder in the control room. I’d come half an hour earlier and record another 15-minute segment in case there was a no-show. I started getting more interested in learning the technology. When I was 12 I wanted to run the records for the classical show. It was very strict, and you had to have your third class licence to do that. The transmitter was expensive and remote-controlled, so you had to learn the 10 steps to turn it on and off, otherwise you could blow it up. You also had to make sure it was set up so you were monitoring the real broadcast coming back. I became interested in the transmitter room up on the hill, but you had to have a Second Class license to go inside. There were some 17-year-olds in there, and I decided I wanted to be part of that; be more than just a disc jockey. That was hard. I spent a whole summer learning all the requirements. You had to be able to draw an FM radio with a heterodyne, name all the parts and describe what they did. And questions like, if you’re charging a lead-acid battery or discharging the battery, when does it give off hydrogen gas? I just memorised the whole SEC book. I figured if I knew how everything worked, I’d be able to pass the test. I passed the test, which was considered a real milestone. It was really exciting that I was part of the group of people that had permission to go in. Those 50,000W transmitters are really powerful. They would vaporise little animals touching it, just disappear in a cloud of dust. But I hardly even wanted to go look at it, I just wasn’t that interested. The fact you could take a signal like a record, send it all over California and recover it with less distortion than a record was what fascinated me. AM was already established. The whole idea was FM would do high quality classical music, and AM would do sports and announcing. Then Governments all around the world put it in, and it spread. To think about the Government getting involved today and making something better… it was exciting to be part of that. Live broadcasts were our thing. We even experimented with early stereo broadcasts. We had condenser microphones which
cost the same as a car at that time, and we were using stereo to create that environment of being at a live show. I don’t know if that was ever completely accomplished, but that was the whole idea. What have we lost with the record compared to the live show? It got me more interested in learning about how we could measure that or build better speakers. That became my passion, to see if we could build things to make that experience more direct. The movie industry did a lot of work on getting the sound better. People complained in the early days of movies that vibrato was ruined and the pianos didn’t sound as good. Walt Disney spent a fortune trying to get Fantasia to sound good. Things like oscillators with Hewlett-Packard to make a panning sound, building and installing the Fantasound multichannel speaker system to re-create the sound of a live orchestra. They had a 100-page magazine article on the work they did. It didn’t make any money for the time, but it was a milestone. Those early days gave me an idea of why I wanted to continue on that path. When we first started working on Constellation to make concert halls, we did experiments in Switzerland. This was in the ’70s, we miked all the instruments and then transmitted it out with a combination of speakers on the stage and more speakers to broadcast it. We realised right away that a close-miked violin doesn’t sound like a violin. Moving around close to a violin yields lots of variation, while moving 20-30 feet away creates a more homogeneous feel. All the instruments were really different in the near-field. After a year-and-a-half in Switzerland we realised we’d have to re-think it and make it more realistic. That led to what we liked about the idea of Constellation. Instead of picking up the direct sound and trying to re-broadcast it, what you do is pick up everything and recreate early reflections; let the direct sound come, directly. Then we could kill the room sound and use high-powered computers to create a cathedral or concert hall. We studied 50 concert halls around the world. We went and measured them very carefully, not just from the stage. We wanted to know what it was like in the balcony, under the balcony, so we’d have a real map of the place. It would take us a week to measure a concert hall so players would recognise the reproduction. It’s a perfect fit for Dolby’s Atmos. They’ve created the solution of objects and we’re going to create change in the room. If they want a helicopter back there, they put a speaker back there to create the sound. We’re about creating the space; going from a submarine that’s reverberant and awful, but when you stick your head out, it’s just like you’re outside.
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