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NT-USB Mini
STUDIO-QUALITY USB MICROPHONE
The NT-USB Mini is a studio-quality USB microphone designed for recording direct to a computer or tablet. Delivering rich tone and boasting handy features like an in-built pop lter and 360-degree swing mount, it’s perfect for podcasting, as well as recording vocals and instruments, live streaming and gaming, voice calls and more. The included desk stand features a magnetic base that can be detached for easy mounting on mic stands or studio arms, and with its simple controls and zero-latency headphone monitoring, the NT-USB Mini delivers incredible audio for all recording applications.
The Cho oice off Toda ay’s Crea ative Ge ene erattion.™ ro ode e.ccom AT 2
Editorial Director Christopher Holder chris@audiotechnology.com.au Publisher Philip Spencer philip@alchemedia.com.au Assistant Editor Preshan John preshan@alchemedia.com.au Art Direction Dominic Carey dominic@alchemedia.com.au
Regular Contributors
Martin Walker Paul Tingen Brad Watts Greg Walker Andy Szikla Andrew Bencina Jason Hearn Mark Davie Mark Woods Jonathan Burnside Rob Holder Greg Simmons
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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 PO Box 295, Ballarat VIC 3353, Australia.
All material in this magazine is copyright Š 2020 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. 25/03/2020.
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The all-new Apogee Symphony Desktop 10 x 14 audio interface blends the professional-grade performance of Symphony I/O MkII with the simplicity and portability of the seminal Apogee Duet to deliver, arguably, the ultimate desktop audio interface. Featuring flagship 24-bit/192k conversion and mic preamps, a hi-res touchscreen interface with single knob control and hardware DSP with Apogee FX plug-ins (including Symphony Reverb) and mic pre modelling, Symphony Desktop is ideal for the discerning professional artist, producer and engineer looking to give their studio the Apogee advantage. Thanks to Amber Technology for the amazing prize! (www.ambertech.com.au)
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COVER STORY
Interface Faceoff: NI KA6, Focusrite Scarlett 2i2, Presonus Studio 24C
24
ISSUE 63 CONTENTS
14
John Storyk & The Electric Lady
Korg Minilogue xd 32 Polyphonic Analogue Synth
Antelope Audio Orion 32+ Gen3 MADI & USB3 Interface AT 6
Mixing Synth Pop Drums: The Process
36
Audio-Technica 3000 Series Wireless
20
28
We’ve just unboxed a new addition to our family Hello HARMAN Professional Solutions!
MadisonAV appointed the Australian distributor of HARMAN Professional Solutions. We’re thrilled to be the newly appointed distributor of AMX, SVSI, and install ranges of AKG, BSS, Crown and JBL Professional in Australia. So in the spirit of great changes, Madison Technologies’ dedicated AV business will now be trading as MadisonAV. Sales Enquiries 1800 00 77 80 www.madisonav.com.au AT 7
NEWS
INTO INTERFACES? SSL continues to peddle its analogue wares to the masses, this year going beyond the SiX and its true SSL “sound in a shoebox” approach and doing something even more accessible: the SSL 2 & 2+. Both are USB bus-powered, desktop audio interfaces with a distinctive look and feel, offering SSL mic preamps and legacy 4K analogue enhancement mode on each inspired by 4000 Series consoles. The 2-in/2-out SSL 2 packs two pres together with 24-bit/192k conversion, a single high-grade headphone output, easy-to-use monitor mix control for low-latency monitoring, and balanced monitor outputs. The 2+ adds onto that MIDI I/O, a second headphone out with its own send, and unbalanced outputs for DJs. Will they compete or SSLip to the wayside? Apogee brings its top-notch conversion into a desktop format with the Symphony Desktop offering 10-in/14-out I/O and all the trappings of the larger Symphony I/O MkII. This new model features Apogee Alloy: a preamp emulation creating “the richest, most authentic audio modelling available in an audio interface”. It models classic American and British tube and solid state sounds. Hardware tracking FX are also included, for dialling in EQ, compression and saturation. Print and Dual Path link modes utilise Symphony Desktop’s hardware DSP processing to provide zero-latency monitoring when tracking. Dual Path Link mode adds unique flexibility, allowing FX adjustment on the track later in the mix, eliminating the need to commit to printing FX when recording. Logic Pro X will offer full hardware/software integration with these features. Presonus debuted its ioStation 24c, a unique mix of monitor controller, interface and DAW controller, amounting to what you might call a studio centrepiece. It’s firstly a 2-in/2-out interface, featuring XMAX preamps, 24bit/192kHz conversion, a high-powered headphone amp with direct monitoring mix knob, and standard line out for monitors. It’s DAW control will work best with StudioOne - Session Navigator allowing editing and automation of projects. Its also compatible with Logic Pro X, Cubase, Ableton Live, and Pro Tools, all of which can take advantage of the touch-sensitive 100mm, motorised fader, a real asset for automation. Also introduced was the Quantum 2626, a rackmount Thunderbolt interface featuring eight XMAX preamps with two direct preamp outputs and two line returns for connection to outboard. Eight line outputs let users run additional pairs of studio monitors or send multiple mixes to musos while tracking. Both 1/4-inch headphone outputs have a dedicated volume control. I/O is expandable with dual ADAT optical and S/PDIF, taking the count to a maximum of 26 ins and 26 outs simultaneously. Audient announced EVO, its first true step into the entry-level space, aimed at the podcasting, streaming and gaming market. EVO 4 and 8 have two and four preamps respectively, offering respectable 58dB gain range. All are SmartGain enabled – gain is automatically set depending on the source, with manual control available to those who want it. Conversion doesn’t reach the industry standard of 192kHz, instead topping out at 24-bit/96kHz, but maybe, just maybe, that’s enough. In a curious move, Steinberg converted a Thunderbolt interface to USB3.1, the AXR4U. A boon for those lacking Thunderbolt, it offers high-res 32-bit/384k conversion and 28x24 channel I/O. Though with Thunderbolt now available on AMD-based PCs and becoming more standard, and with USB3’s patchy history of driver and OS interface implementation, it remains to be seen if USB3 will catch on in a big way. IK Multimedia showed off its iRig Pro Duo, a powerful mobile recording solution, with two phantom-capable combo inputs, full bus-power off mobile devices and MIDI I/O. Certainly seems a lot for an iOS focused interface, but its improved PC drivers may make it an outside option for home users as well.
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MORE NEWS AT www.audiotechnology.com
Amber Technology (SSL, Apogee): sales@ambertech.com.au Link Audio (Presonus): sales@linkaudio.com.au Studio Connections (Audient): sales@studioconnections.com.au Yamaha Australia (Steinberg): 1800 805 413 Sound+Music (IK Multimedia): sound-music.com
M50X
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Long heralded by online reviewers and top audio professionals, and backed by a cult-like following of serious music fans, the ATH-M50x offers an unmatched combination of audio and build quality for exceptional performance both in the studio and beyond.
For information on the full range of M-Series headphones go to audio-technica.com.au AT 9
LIVE NEWS
LIVE d&b‘s new KSL-SUB and KSL-GSUB operate with two forward-facing 15-inch drivers and a single, rear-facing 15-inch driver. The sleek sub weighs in at 82kg, measures 1 x 0.90 x 0.45m, and boasts an output of 139dB when driven by d&b D80 amplifiers. A new KSL-SUB adapter frame allows the KSL-SUB to be hung mounted above KSL top boxes in a flown line array. The KSL-GSUB has a more streamlined form without the standard rigging hardware of the KSL-SUB. Its 45cm height allows for under-stage positioning. The dBTechnologies VIO C Series is a flexible two-way active line source designed for ease of rigging and diversity of coverage options. The C12 and C15 are single woofer designs (12- and 15-inch), and the C212 features two 12-inch woofers. The whole series is equipped with latest gen Digipro G4 1600W RMS Class D Amp module, driving powerful DSP and allowing each system to deliver impressive SPL (139dB for VIO C12, 140dB for VIO C15, and 141dB for VIO C212). The 22.5° coverage is standard across the range, and cabinets can be vertically coupled or set up side by side, all controlled via RDNet cards and Aurora Net control software. More Quantum: Digico released the 338 and Quantum 5. The Quantum 5 takes the shape of a scaled down Quantum 7, similarly designed to be a drop-in upgrade to the SD5 console. Quantum 5 expands the SD5 to over 450 channels of processing at 96k, and equips the console with MADI ports and full DMI card support. In software, Nodal Processing allows for flexible and creative IEM mixes, True Solo for monitoring them, 48 channels of Mustard processing and 12 Spice Rack processing slots add flavour, including Chili 6. Mustard channels comprise tasty channel strips with preamp models, EQ, compressors and a gate which can be used alongside standard Quantum processing. Spice Rack functions more as a FPGA-based plug-in collection. L-Acoustics‘ Kara II features Panflex, which gives Kara four-in-one directivity, including: 70° or 110° symmetrical and 90° asymmetrical, steering to either the left or right. In its 70° configuration, Kara II packs a full 2dB more than in 110°. L-Acoustics will make available a kit to upgrade existing Kara with Panflex, for increased return on investment. The 24kg active boxes are a two-way active WST design, span a 55Hz – 20kHz frequency range and give 142 dB maximum SPL. The PXM-12MP marks Electro-Voice’s entry into the powered stage monitor market. Created for live performances from solo acts, bands, mobile DJs and rental companies, PXM-12MP is a two-way stage wedge featuring coaxially aligned 12-inch LF and 1.75-inch HF transducers. Its small footprint and light weight of 13.5kg makes it an extremely portable solution suited to live performances on all stages, particularly excelling where floor space is limited. Mackie SRM V-Class was unveiled: Mackie’s claimed “reimagination” of the powered monitor segment. The SRM210 V-Class, SRM212 V-Class, and SRM215 V-Class, with 10-, 12-, and 15-inch low-frequency transducers, respectively, all run new Class D 2000W amps. All have dispersion areas of 60° (H) by 40° (V), a four-channel digital mixer with two channels equipped with a mic/line/instrument combo input, a dedicated 1/8-inch stereo aux input, and stereo Bluetooth input. In addition, two SRM V-Class loudspeakers can be wirelessly linked, with up to 100 meters of range between linked speakers. Linked speakers can operate in stereo or as dual zones. The Evolve 30M Column Speaker is designed to deliver room-filling coverage and flexible functionality combined in a very compact package. The system includes a speaker array, a two-piece pole connector, an array and pole backpack carry case, and a powered subwoofer, all made to pack up and assemble quickly.The full-range column array and its six 2.8-inch drivers provide ultra-wide 120° coverage via proprietary waveguides; array-formed 40° asymmetrical vertical coverage ensures acoustic output is directed towards both sitting and standing audience members, with 1000W of power shared between it and the sub. It also packs a six-channel mixer with DSP.
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MORE NEWS AT www.audiotechnology.com
NAS (d&b, db Technologies): sales@nas.solutions Group Technologies (Digico): sales@grouptechnologies.com.au Jands (L-Acoustics, EV): sales@jands.com.au Amber Technology (Mackie): 1800 251 367
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RECORDING NEWS
NAMM STUDIO LUNA is Universal Audio’s surprising new DAW outing. It transforms UA’s interfaces into fully integrated production systems, using its integrated DSP to create an analogue-style (Mac only) DAW centred around latency-free processing. LUNA offers precisely emulated audio summing circuitry from the Neve 80-Series mixing consoles. Neve Summing is a LUNA Extension directly built into LUNA’s mixer, adding instant colour. Other Extensions include a Studer A800 Multitrack Tape. LUNA also sees UA’s first foray into software instruments (Minimoog, Steinway Model B and other instruments included). Available free for Apollo and Arrow Thunderbolt owners. Moog has released its Sub Phatty successor — the Subsequent 25. Subsequent 25 has upgraded from Sub Phatty with the implementation of wood side pieces, twice the headroom for access to a new range of tones in mono and duo modes, and an upgraded keybed for improved playability. Moog has reshaped the gain staging in the Ladder filter to boost harmonic saturation and analogue compression, as well as re-tuning the multidrive circuit and increasing the power of its headphone amplifier. Sounds like a sweet modern update to a mono classic. Still breathless from the hype of the OC818 mic, Austrian Audio slipped out a headphone reveal, the Hi-X (High Excursion) Headphones. The onear, supra-aural HI-X50s are specifically designed for those requiring a portable and compact option, whereas the HI-X55 circumaurals provide over-ear comfort. Both feature Austrian Audio’s exclusive 44mm HI-X driver and have a 250Ω impedance. Austrian Audio’s ring magnet system is present in the new headphone line, further enhanced by improved airflow and the strongest magnetic field the company has made yet. Things are coming full circle for MXL, which announced the development of Revelation II, a follow up to the brand’s first-gen condenser. The new mic will offer the warmth and intimacy of a quality tube microphone, but with extended clarity and punch for balanced recording. A dual gold-sputtered, 6-micron diaphragm and hand-selected EF86 pentode tube will provide warm, rich and transparent sound. The Revelation II’s polar pattern selection is virtually limitless, controllable by a variable pattern control knob. AMS Neve has packed lunch – the AMS RMX16 Digital Reverberation System in 500 series format. Designed to deliver all the musicality of its famous predecessor, but at a ‘fraction of the price’, this new unit incorporates the nine programs that came as standard with the original AMS RMX16, as well as the nine rare aftermarket programs that were only available to users via a remote control with barcode reader input. RME announced three new AVB and MADI products, the 12Mic, the small AVB Tool and the mammoth M-1610 Pro. 12Mic features 12-channel microphone and line level inputs with digital studio quality converters, remote controllable gain, integrated MADI and AVB connectivity. A web front-end offers access to the device controls and its integrated 268×282 channel routing matrix. AVB Tool offers similar capabilities with four preamps and a 260x260 channel router. The M-1610 Pro brings plenty of analogue I/O to any studio setup, integrating 16 analogue inputs with switchable sensitivity per channel of up to +24dBu, eight corresponding analogue outputs, and an additional headphone output. In controller news, Avid released the S1 Control Surface, the smallest in the S series, offering all the power of the S6 but for home studios and smaller projects. It’ll hook up to an iPad for visual feedback of your Pro Tools and Media Composer control, and will hook up with other S1s for an expandable surface. Novation released the Launchpad Pro MkIII, which improves on its predecessor with a built-in, four-track 32-step polyphonic sequencer for controlling hardware via its new MIDI I/O, as well as other playability and connectivity improvements. Mackie expanded its CR series of studio monitors with 3-, 4- and 5-inch models, as well as Bluetooth-equipped editions of the entire line. Joining them is a new subwoofer, the CR8s-XBT, a 200W Bluetooth sub for extending the bottom end of its nearfield brethren.
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MORE NEWS AT www.audiotechnology.com
CMI (UA): sales@cmi.com.au Innovative Music (Moog, RME, MXL): info@innovativemusic.com.au Group Technologies (Austrian Audio): sales@gtaust.com Audio Chocolate (AMS Neve): audiochocolate.com.au Avid: avid.com
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FEATURE
It’s been 50 years since John Storyk first designed Electric Lady Studios for Jimi Hendrix. With over half a century of skin in the game, Storyk talks about what’s changed and what hasn’t in studio design. Story: Mark Davie
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“I have a drawing dated February, 1969,” said John Storyk. It’s the earliest memento he can find of arguably his most famous commission; the Jimi-Hendrix-owned Electric Lady Studios in New York. The drawing doesn’t actually show a studio, rather it’s a redesign of the club, Generation, that existed on the Eighth Street premises at the time. Hendrix was a frequent performer and patron of the club, and — perhaps because of his rising bar tabs — decided to buy the lease with his manager, Michael Jefferey. Storyk knew Generation well. A year earlier — on the day he finished college — he’d driven to Manhattan in search of his own musical fortune. He played sax and keys for a Blues band styled in The Commitments’ likeness. “We’d listen to Blues guys like Junior Wells, James Cotton and Muddy Waters at a club in New York,” he explained. The club was Generation. Thus began a long list of events that would spell Serendipity for the young Mr. Storyk. While music wasn’t paying the bills, Storyk had to get a day job and ended up in an architecture office, which did not push his creativity to the limit. Searching for something that sounded more fun, and seeing an ad one evening while being asked to wait an extra few minutes on line for an ice cream, he answered that ad, requesting free carpentry labour for an experimental night club, downtown. It was the Summer of ’68 — “Yes, I did inhale” — when young artists like Storyk were inclined to follow their nose, rather than their wallet. He met with the owners and volunteered his labour on the condition he could redesign the club. THINKING IT THROUGH
That club became Cerebrum. A 64-person capacity venue in New York that was a cross between a club and theatre. Not Broadway’s kind of theatre; experimental, psychedelic, dress in a white robe and dine on marshmallows, ‘you are the show’ kind of theatre. It lasted nine months, but had a heavy cultural impact, especially on Jimi Hendrix who was looking for an architect to overhaul Generation. “It was the vibe he wanted,” said Storyk, who at 22, “got a call from Jimi Hendrix to do a club. Jimi wanted everything to have curves, and white walls with changing lights. He had this idea that at the back of the club would be a control room that could record the music played in the club. It doesn’t sound particularly complicated today. But in 1968, that sort of thing wasn’t happening.” At the last minute, Jimi’s long-time engineer Eddie Kramer, convinced him he shouldn’t build a club. Instead, he should funnel the $3-400,000 a year he was spending on studio time to build his own. “I watched my first commission arrive and disappear in a matter of weeks,” said Storyk. However, Jimi and Eddie suggested he stay on and tackle the studio. Storyk was open about his inexperience, but they had confidence he’d figure it out. “I quit my architecture job, visited all six of the studios in New York at the time, and hired a young Phil Ramone to give us advice,” said Storyk. He and Kramer plowed through the project, scouring the scant resources available. He even offered his services free of charge to an acoustician who specialised in radio
stations so he could learn how to draw and create architectural isolation details — super required for the studio’s infrastructure. 15 months later Electric Lady studios opened. Storyk adds “…these were amazing days — drawing, building the studio in the day, continuing my graduate education at night at Columbia, studying with Buckminster Fuller down the street during the summer of 69; Woodstock Festival; music everywhere….”
It’s not Jimi Hendrix’s spirit in the walls, the coloured lighting, the fact you have to slide in a tiny door on Eighth Street, or the river that flows under the studio. All interesting stories, but the science is in the ceiling
LADY LUCK
“It wasn’t until many years later that I realised we got a little lucky, especially with the ceiling,” said Storyk. “I went back and found out why the room worked so well, and it’s because of the ceiling, which looks like a flat propeller. It was partly aesthetics, and partly a half-baked notion that I shouldn’t have a parallel ceiling. I also remembered reading a little bit about membrane absorbers. They’d been used at RCA Studios and the great Kultura Studios in Moscow. I didn’t completely understand the science and took a stab at how to build it. We used very lightweight plaster and the ceiling turned out to be a giant membrane absorber, which keeps the low frequency reverb time down in the room.” Storyk was frank: “I’m not going to lie. I had very little science to back up what I was conceiving, but Eddie and I took a shot at it and it worked out.” It wasn’t until many years later that he realised it gave Electric Lady a desirable large studio characteristic of tailing down the low frequency reverb time. It tightens up the sound, as opposed to concert halls, where you want it to tail up and bloom. “That’s what’s doing it,” said Storyk about the studio’s ‘sound’. “It’s not Jimi Hendrix’s spirit in the walls, the coloured lighting, the fact you have to slide in a tiny door on Eighth Street, or the river that flows under the studio. All interesting stories, but the science is in the ceiling… although we all still wonder if Jimi is still with us sometimes” STUDIO PODS
These days, Storyk’s company, WSDG, that he started with his wife Beth Walters (the W) is averaging about 40-50 studios a year. In its
lifetime, WSDG has worked on thousands, along with halls, theatres, AV design, conference rooms and structural acoustics, “but the studios are still what I love”. Lately, those studios have been getting smaller and many are specifically geared towards podcasting. ‘Small format streaming facilities’ have become more and more common. WSDG has been designing spaces for all the big players; Audible (owned by Amazon), studios in LA and NY for Midroll (owned by Scripps), and a huge facility for Gimlet, which got bought out by Spotify for US$200m this year. Every radio station is essentially in the streaming game now, as podcasting on demand has turned breakfast and drive-time shows into ‘anytime shows’. The variety of content and production styles could mean anything from a one-person/ no-engineer facility to a four-person, fully-staffed interview setup. “In the case of Gimlet, a 45-minute Gimlet podcast could take months to make,” said Storyk. “It could have been done in 20 different segments with live people, phone interviews, and purchased material woven into the product.” Often the equipment is kluged together from off-the-shelf products not specifically tailored to podcasting, and part of WSDG’s job is to piece together high-end audio systems that can integrate phones — “not as easy as it sounds”— and be operated by a podcast producer with no engineering experience. When he saw Rode’s latest product, the Rodecaster Pro, Storyk was impressed and wondered why it had taken so long for a manufacturer to pull all those pieces together into a podcasting-specific console. A lot of the studios also have intense isolation requirements. Gimlet’s facility features 10 tightlypacked studios, and because most of these top end facilities are in Brooklyn, Manhattan or Los Angeles, no space goes to waste. Admittedly, says Storyk, the majority of podcasting is centred around the voice, so the low frequency isolation requirements are often reduced. However, the smaller the room, the higher the frequency of the first order mode, which can often land in the speech range. Small rooms are often more complex to deal with than larger ones”.“ While you’d be forgiven for thinking a company like WSDG would only spec, bespoke, customdesigned acoustic treatment, Storyk is a fan of offthe-shelf products [see Beating the Low Frequency Trap]. He also lauds the ability that today’s modern iPhone can make 90% of the acoustic measurements we need. This is amazing! That means thousands of people can make measurements, which takes the voodoo out, and puts a little more science back into it.” Unlike some acousticians who would be happy to maintain the perception of their practice being a dark art, Storyk is pleased about the dissemination of measuring tools and the information available about acoustics. “People are getting smarter,” and Storyk applauds that. He would have killed for it when he was designing Electric Lady. MARKETABLE ACOUSTICS
With over 3000 studio rooms under his belt, Storyk admits there have been times over the years when he has worried his interest in the subject would wane. Always one for the science as well as the AT 15
BEATING THE LOW FREQUENCY TRAP Storyk believes there have been three major changes in control room design over the years. “Firstly, people are more interested in low frequency than ever before, both because of the nature of the music we listen to and the advancement in the delivery system technology.” Thankfully, the second change has been in the development of off-the-shelf acoustic products, specifically thin ones, that can control low frequencies. The third major change is the size of control rooms. Over the last 10 years control rooms have been getting smaller; often because they only house single
users, and real estate is more expensive. “There will always be what Chris Stone (legendary Record Plant co-founder) called Mothership studios like Electric Lady, Abbey Road and Capitol, but we’re clearly not building a lot of those rooms anymore. Instead we’re building thousands of smaller rooms where we listen to more low frequency information than ever before.” While a typical studio in the late ’80s or ’90s would have big corner traps with “lots of fuzzy stuff and hanging baffles inside them”, they also had the room volume to cater for them. That approach doesn’t work so well in a room measuring three by four metres. “I
can’t be having a one metre deep bass trap in a room with only two and a half metres of headroom.” The ability to buy pre-fabricated treatments off-theshelf that control low frequencies within four inches or less in depth, is a game changer. Pressure absorbers — resonators or membranes — can be tricky to design. Then you have to take your chances as to whether you made them correctly so they perform to spec. These days, you can buy a product the manufacturer has tested and can reliably produce to spec.
THE SOUND OF JAZZ
Stitcher Studios’ control room with a view into the live room. (Photo: John Muggenborg.)
artistry, the arrival of Time Delay Spectrometry (TDS) was one of those developments that piqued his interest along the way. TDS helped birth some of the industry’s most ‘marketable’ studio designs, and unravel others. “Live End Dead End came about when we had TDS and specifically when we had measurement devices that for the first time gave us data that mirrored what we were hearing. “Everybody realised that certain treatments were at the wrong end of the room. Chips Davis said, ‘Why don’t we put the dead end in the front, and the live end in the back?’ Peter D’Antonio then said, ‘We should put diffusion in the back.’ Which is a sharp idea. Then the acronym came about and it got marketed.” Nothing wrong so far, except that it makes “little sense for the low end. It made no sense to put all your low end absorption in the front, and low end diffusion or reflection in the back.” Storyk reckons it should have been called Frequency Dependent Live End Dead End, “but six-letter acronyms don’t roll off the tongue.” Similarly, Storyk says the Westlake design AT 16
developed by Tom Hidley “had some fallacies”. Again, the TDS analysis proved revealing. “We were now able to see that the compression ceilings were a mistake.” The original intention of coupling a hard ceiling to a baffled hard front with big Augspurger-style speakers was to elicit more low end. Great news on one hand. The bad news was you would hear a first order reflection off the ceiling, creating disturbing comb filtering as energy bounced off the consoles. “It often occurred to me that you could have gotten more low end by just adding a sub,” said Storyk. “But people weren’t up to that.” This isn’t criticism from the outside. In the early ’70s, Storyk and Hidley designed a room together for Stevie Wonder, which had a compression ceiling. It was Studio B, a small Quad room at Record Plant. “Stevie lived in it for two years with his engineers/producers Bobby Margouleff and Malcolm Cecil. One day, I walked in to one of the sessions, and there was a packing blanket covering the entire console. Bob had his hands underneath
One of WSDG’s most prominent commissions was its work for Jazz at Lincoln Centre (JALC). The organisation had two firms in mind for the development — Russell Johnson’s (“a brilliant acoustician who knew more about large halls”) Artec, and WSDG — but they wanted skills from both companies. Over one weekend, the two companies got into a room together and hammered out a partnership arrangement that existed solely for that one project, they called it Sound of Jazz. JALC has three performance spaces; the 1230-capacity Rose Theatre, the 420-seat Appel Room, and the intimate Dizzy’s Club. Arguably the most challenging space is the Appel Room, being the most prominent from the exterior, with a 50 by 80-foot wall of glass serving as the stage backdrop and overlooking New York city’s spectacular Central Park and Manhattan skyline. Obscuring the multi-million-dollar view wasn’t an option, but the glass presented a massive problem. The acoustic team solved the reflection conundrum “by angling the inner glass to direct all the acoustic information up to the ceiling,” explained Storyk. “Then we made the swooping ceiling out of tubes, with a lot of air between them. The ceiling doesn’t look transparent, but acoustically, it is. All the energy goes up through the ceiling, and on the underside of the infrastructure — black-painted catwalks and ducts — are hundreds of cheap RPG 2D Styrofoam diffusors. They sprinkle the sound back down into the room, which is what you want from an acoustic jazz environment.”
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EXPERIMENTAL CONTROL “How many pieces of information does someone have to give you in order to determine the nature of a car?” Storyk asked. “0-60mlph time? Braking distance? Number of doors? Horsepower of the engine?” It turns out you need quite a few. It’s the same with studios; no single characteristic describes a studio. While reverb time is an extremely good descriptor of live rooms — especially large ones — it’s not the only characteristic. It’s usefulness is also aligned with the fact it’s relatively easy to measure, and we’ve been measuring it for many years, posits Storyk, making it easy to compare. Another characteristic of studios is quietness, lists Storyk: “On the control room side, reverb time
carries even less weight. Low frequency modal behaviour is important: where and what are the modes and how are they spaced? “Accurate frequency response from the monitors at the listening position is an obvious one. What’s happening in the time domain — phase — is critical. And I want to see first order reflections; where those console bounce points are. Are there any reflections inside the initial time delay gap (ITDG)? The concept of ITDG is more complicated, and doesn’t sit neatly within an acronym, so people don’t like to talk about it. “The idea is to understand what’s happening at the listening position when you compare the direct sound to the next set of reflections. If
you have those pieces of information arriving inside a 10dB difference, between 15 and 30ms, then you risk a comb filter problem. The obvious example of this seen in many studios is a console bounce, but it could also be a piece of equipment or a low ceiling. “You can connect the dots. As the rooms get smaller, the surfaces get closer and this phenomenon gets more hazardous. It has nothing to do with reverb time, because if you plug the dimensions of a small control room into the Sabine equation, it is essentially dead because of its size. In small rooms, often individual surface geometries have little relationship to reverb time.”
It should have been called Frequency Dependent Live End Dead End, but sixletter acronyms don’t roll off the tongue
it while he mixed on the knobs and faders. “Bob said, ‘I don’t know what it is, but it’s better when I put the blanket over the console.’ Of course, the speakers up high were bouncing off the console, then off the ceiling, back off the console and back up again. He was essentially hearing a comb filter. He didn’t put the science to it, neither did I at the time, but his ears knew it.” These days, one of the technological developments that most excites Storyk is Auralization. In short, the ability to listen to architectural drawings. “It’s pretty common now,” said Storyk. “Most acousticians will use it, though there’s only a handful of software manufacturers, and it’s not easy to use. We looked at the most popular ones (CATT, EASE, Odeon) and at the moment are still most comfortable with CATTAcoustic.” WSDG has Auralization rooms in both its Swiss, Berlin and New York offices. It allows designers to model acoustic environments and then listen to them in a calibrated room. “It’s one thing to read graphs, it’s another to listen.” WHITE, WITH COLOUR
Storyk feels indebted to Electric Lady Studios. “Career tip: Make your first project famous!” He said. “Before it opened I had four more studios to design.” AT 18
At the time, Storyk could never have known the impact Electric Lady would have. Not just on the New York recording scene, but on the entire recording industry. It was one of, if not, the first artist-owned studio. Jimi’s plan was unheard of; he would dictate the record making cycle to the label. “He was going to make his own records, at his own pace, then when he was done, he would hand them to the label. Few artists were doing that.” The studio was designed for artists, who were often seen as “the unnecessary people on that side of the glass,” said Storyk. “It didn’t matter how they felt; the quicker they got out, the better.” It meant he designed a larger control room, to make more room for artists on both sides of the recording process. Jimi also wanted the furniture to be part of the mood, to make the live room feel like a living room. Coming from Cerebrum and then the shelved Generation re-design, Storyk simply figured recording studios should be architecturally interesting, too. That too was not the norm. One of Jimi’s biggest takeaways from Cerebrum were the ability to shift the mood with colourchanging lights projected onto white walls. Back then, it wasn’t easy. At Cerebrum it was accomplished with 20 Kodak 35mm slide
projectors. At Electric Lady, lighting designer Bob Walsch had to buy quantities of bulbs in three different colours, and attach them to giant dimmers to mix the colours. Four years ago, things came full circle. Paul Epworth had bought The Church Studio in London and wanted to overhaul the design. He’d worked at Ann Mincieli’s Jungle Studios in New York, a WSDG job, and wanted that specific Augspurger monitor system. It turned out one of the rooms at the The Church was almost the same size as the Jungle Studios control room, which worked out nicely. “In one of the conversations with Paul, we asked what he felt it should look and feel like,” said Storyk. “He said, ‘I’ve got this idea where everything’s white, and we just change the colours and the mood with lighting.’ I thought to myself, ‘I haven’t gone very far in 45 years!’ “I didn’t tell him that though, I just said, ‘That’s a good idea… I think we know how to do that.’”
M A K E YO U R M A R K
305P MkII 306P MkII 308P MkII
Detailed Imaging | Wide Sweet Spot | Superior Accuracy · Image Control Waveguide originally designed for the M2 master reference monitor · 1” Neodymium tweeter with superior transient response
· Refined low-frequency transducer for improved linearity · Deep, accurate bass via a patented Slip Stream™ low-frequency port
· Powerful, integrated dual Class-D custom amplifiers · Boundary EQ and HF Trim tailors performance to your environment
P R O U D LY D I S T R I B U T E D I N A U S T R A L I A B Y C M I M U S I C & A U D I O
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CMI.COM.AU AT 19
TUTORIAL
In Part One of this series about mixing synth pop, mixer/engineer Tristan Hoogland describes his process for mixing drums. Tutorial: Tristan Hoogland
Having started out working in the pop and rock genres with guitar bands such as Ball Park Music, Fountaineer and many others, my first foray into the synth pop and electronic music realm was anything but smooth. The usual tricks and techniques I used for pop and rock music no longer worked, and it took a while to appreciate that the sonic palette was completely different and to align my hearing accordingly. I hope this series provides some insights into the general philosophies and techniques that go into making this music pop! DRUMS: LAYER UPON LAYER
In synth pop the drum elements are often comprised of various individual layers. It’s not uncommon to receive numerous kicks, snares, claps, fingersnaps and hi-hats – not to mention anything else that may exist. Just like having multiple mics on a single source (e.g. a kick in and a kick out), each track contributes a small part to the final sound and it’s our job to make them all work together. In terms of the overall presentation of the drums, the kick is generally quite prominent with a focus in the low end and low mid area; think in terms of ‘knock’ and ‘thud’. Snares, claps, and fingersnaps tend to be ‘crispy’ rather than ‘fat’ and ‘cracky’, and hats are ‘thin’ rather than ‘chunky’ and ‘cutting’. AT 20
It all depends on the arrangement, of course, but broadly speaking I’ve found the above to be true. IT’S ALL ABOUT THE BALANCE
Start by pushing up the faders one by one, keeping a mental note of anything you might notice as you stack up the sounds. I’m usually keeping an eye out for anything that’s bloating the low end with the kicks, and anything that’s making the upper mids feel harsh with the snares, claps, percussion and hi-hats. Your first impression is usually right; if you feel like there’s anything you can address straight away that may be a problem later, I’d encourage you to deal with it then and there. PHASE & TIMING
I spend a lot of time tinkering with the polarity and timing of the samples. Don’t just import stems then stack ‘em up and go, because it’s not uncommon for similar elements to audibly arrive later or earlier than one another. It might be that a sample has a 2ms gap at the start, or it was played in live with natural timing variations.
Start by identifying the main layer of each sound and get it in positive phase (meaning that the initial transient moves upwards/positively), then manually align the remaining tracks by hand. Next, establish a coherent phase relationship. One by one invert the phase of each track against the main
layer, listening for the setting that provides the most focused sound. For advanced polarity correction I use UAD Little Labs IBP Phase Alignment Tool, which continuously varies the phase between 0 and 180 degrees. Aligning the tracks, adjusting the phase and/or inverting the polarity can make for a night-and-day difference. Be sure to use your discretion when aligning tracks, however, as it may be the producer’s intention for them to land slightly differently. EQ: THE ART OF CRAFT
EQ is a powerful crafting tool when attempting to make multiple sound elements work together as one. I tend to reach for subtractive EQ rather than additive EQ, with the goal of making space in one element for the other elements – especially when
there are so many to fit in. Don’t be alarmed if you find yourself getting heavy handed; the more layers you have in a sound, the more EQ you’re likely to use to make them all fit. I’ll start with a medium to narrow Q and sweep through the areas of a track that I feel have excessive energy and are masking other tracks or distracting me from what I want to hear most in the sound, and reduce those areas accordingly. KICKS: SCHOOL OF HARD KNOCKS
Kicks in synth pop music tend to favour ‘knock’ over click and sub, so I’m usually looking to reduce ‘pointiness’ (2kHz to 5kHz) and careful not to take away too much low mids (180Hz to 400Hz). In the example shown here I’m reducing 4.21kHz by -10.8dB and 715Hz by -4.3dB (bottom right), with a +4.2dB shelving boost at 98.7Hz on the kick
buss (top left). Clickiness in the upper mids can usually be removed using a very tight Q without significantly altering the overall tone. You’ll notice I’m not adding any upper mids or top end on these EQs because I’m wanting to focus on the low end and low mids to get that ‘knock’. I’ll do quite a lot of this work first on some smaller speakers, like Auratones or NS10s, and then switch over to the main monitors to make sure I’m not overlooking anything in the low end. Some samples will have a lot of resonance and bloom that will muddy things up or be out of tune with one another. I’ll experiment with high pass filters, sweeping all the way up to 200Hz on each sample just to see how much clearer I can get the low end, and then start parking it in a more reasonable spot (anywhere between 30Hz and 120Hz). Tip: FabFilter’s Q3 can be good for helping you ‘see’ the low end, and also for overlaying each track to see if you’re getting any excessive overlap between samples. Another handy tool for shaping the low end is a transient designer. My favourite is iZotope’s Neutron Transient Shaper. The multiband option allows you to tighten the low end without affecting the rest of the spectrum. Setting the crossover between 80Hz and 180Hz can really help fine tune the ‘knock’ in the above band without adding a lot of crazy sub and bloating the low end. SNARES, CLAPS & FINGERSNAPS
Snares and claps tend to build up in the mids and upper mids, resulting in unwanted honkiness and harshness. I spend a lot of time managing two areas in particular: 600Hz to 1kHz, and 2kHz to 5kHz. I rarely boost, except to add a top end bump if I feel like the cuts are creating a loss of overall presence. One of my favourite techniques is using Lo-Fi (part of the D-Fi suite in Pro Tools) on uninspiring or ‘pointy’ percussion that lacks body and fullness, such as fingersnaps or claps. Simply wind up the distortion to taste and you’ll be surprised by how full the sound becomes without any other processing. AT 21
HATS
Similar to snares, I find in order to get them all working well together I need to make cuts in the higher end of the spectrum, generally the 3kHz to 6kHz area. I’d also recommend experimenting with Lo-Fi, bringing down the sample rate and antialiasing sliders because they can help make the hats more present in the mids and far less thin and pokey. VINYL SAMPLES, iPHONE RECORDINGS & LO-FI DRUMS
Flavour tracks such as iPhone recordings and vinyl rips of live-played drums often make appearances in electronic music, but they need a lot of work to sit. I aim to have them sit in the mid range and leave the top and low end for the samples. Therefore I’ll start by setting a high pass filter quite high up to remove any unwanted rumble; it may end up as high as 300Hz. Old vinyl and iPhone recordings usually have more noise rather than musical top end, so I’m not afraid to be low passing down to 6kHz or lower if need be. You can automate these settings depending on how dynamic the arrangement is; for example, open them up if the rest of the samples disappear for a few bars. If it’s a complicated stem with a full kit I might use a dynamic EQ or multiband compressor, mostly just to keep the hi hats and snares in check. PARALLEL LINES
One of the struggles I first had mixing electronic drums was getting them to have impact. My tricks from the days of mixing rock bands, such as parallel compression, didn’t cut it – things would just get louder, not bigger. I discovered that introducing distortion in addition to compression would pay dividends. My philosophy is that the compression adds ‘smack’ and keeps the drums solid, while the distortion helps them cut and adds aggression. (As a preface to the following, I rarely use compression on individual tracks. If hits are seemingly inconsistent, and assuming they’re samples, I’ll manually correct them with clip gain or automation. Beyond that, I’ll resort to saturation tools such as Lo-Fi on a case-by-case basis, and achieve the rest with parallel compression.) Here’s how to do it. Start by creating three auxiliaries. The first will be our Drums Submaster, which we’ll route all the drum elements through with zero processing. The second aux will be for parallel AT 22
compression, and the third for parallel distortion. Establish the required start balance by adjusting the amount of each drum element that is sent to the Drums Submaster. Now send each individual track to the second and third auxiliaries respectively and set them to post-fader. Sending them discretely like this offers the freedom to not only drive more or less of each element into the parallel processing, but to have control if any one element is being brought too far forward or not forward enough. Getting the parallel compression to work is straight forward. I usually opt for a Distressor with a medium ratio (around 3:1) along with a fast attack and fast release so it’s working quite hard – I aim to see 10dB to 12dB of gain reduction. The Distressor’s meter is quite reliable, and I expect to see a similar amount of gain reduction on each major transient event. If one or two elements are driving it less or more than the others then I know that my fader balances and/or EQ need revisiting. For the distortion I use Soundtoy’s Devil Loc, or sometimes UAD’s Thermionic Culture Vulture. I always set the Devil Loc to Crush = 3, Crunch = 3, Release = Fast, Darkness = 2.5 and Mix = 10. The Devil Loc is an abusive unit and the trick to getting it to work is to have control of the level you’re driving into it. I’ll either assign a master fader to the bus or place a trim plugin before the Devil Loc so the individual sends can always be parked at 0dB. Starting with the fader or trim control all the way down, slowly turn up the signal going into it until you find the sweet spot – which I usually find
to be quite low! Now blend the returns of the two auxiliaries (parallel compression and parallel distortion) in with the Drums Submaster to taste. If things are feeling too harsh back off the Devil Loc return. I typically end up with a fairly even amount of each, but if the source material is lacklustre I might end up with more Devil Loc. Once you’ve learned how to dial these in – particularly the Devil Loc – you won’t mix drums without them again! EFFECTS
Unless effects are printed with the sample my advice is don’t bother adding them. With electronic music, and especially with drums, you’re going for immediacy. Adding reverb will smear the transients and set things back rather than forward. I can also guarantee that it’ll be the first thing your client asks you to remove! UNTIL NEXT TIME...
Hopefully the techniques mentioned here have armed you with some ideas for your own projects. Try implementing one or two into your existing workflow, such as experimenting with parallel processing or parallel distortion instead of inline processing, and see how it works for you. The palette is always changing, but developing some tools and techniques that create a level of predictability in your workflow is one of the key things in mixing. Besides that, anything goes! Next issue: Mixing bass and synths.
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REVIEW
Interface Face-off
The Presonus Studio 24C, Native Instruments Komplete Audio 6, and Focusrite Scarlett 2i2 may differ in size and I/O, but all are rugged, road-ready and starter-friendly interfaces. Which one best suits you? Review: Rob Holder
PreSonus Studio 24C BEST FOR RECORDING
Focusrite Scarlett 2i2 3rd Gen BEST FOR MIXING
Native Instruments Komplete Audio 6 BEST FOR PRODUCING
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The Komplete Audio 6 is a MkII refresh of its years-old namesake. Now a black, minimal slab of glass and aluminium, it’s a stylish looking unit. It has two preamp-equipped combo jacks, two extra TS inputs on the back (CV-ready for use with modular gear), MIDI in/out, as well as two stereo outputs. One standout addition is the recessed USB-B port, making your computer connection rock solid (it takes some effort to get it in and out). A pair of high-power headphone outs are also included, for the odd collab or sharing a production with a mate. There are VUs for all the ins and the main out, and a big fat volume knob on top. The Presonus box ain’t so pretty, but what it lacks in looks it makes up for in ruggedness. Once again, we get two combo preamp inputs, but no extras out back. It comes with a stereo out, headphone out (on the rear… why?) and MIDI I/O, too. Instead of traditional B, you’ll find slimline USB Type-C on the back, ideal for versatility, making for easy swap outs if you get caught short (the 24C ships with two USB cables: Type C to C and Type C to A). The front panel knobs feel solid, and make accurate incremental adjustments easy with their weight and resistance. The evergreen 2i2 from Focusrite is fresh from its 3rd gen makeover, so it now sports reflective glass front and back I/O plates, and shiny, primo-feeling everything. It’s a very nice unit to behold and to operate, netting you the same I/O as the Studio 24C sans MIDI. Its knobs lack markings for gainmatching (like the Presonus), but all have a nice resistant feel for fine adjustments. Focusrite also graces us with USB-C, another welcome addition. VOCAL SHOOTOUT
preset, I was instantly in the ballpark and after pulling out a little of the saturation and a bit less of the highs, I was happy. Less enamouring was the Lexicon MPX-i reverb, which routinely laid on the reverb with a trowel and needed the Wet signal dialling down to around 10% to be usable. Elsewhere in the supplied suite are the Plugin Alliance-sourced mix tools, with a solid dual-band EQ and transient processor along with the highly capable Brainworx bx_opto. NI’s box of software tricks aren’t tailored for the home studio vocalist so much as the home studio producer. The Solid Bus Comp is the go-to channel strip which I applied to my vocal. Replika was inspiring, offering a super-wide, dispersed delay with minimal effort. My resulting vocal was cool, but not as practical or polished in the mix as the PreSonus one. As mentioned, this isn’t the KA6’s sweetspot. My Scarlett vocal chain used the Focusrite Red3 compressor, which happily took a pounding — sounding great as I turned the thumbscrews. Equally pleasing was the Softube TSAR-1R reverb, giving a bright, expansive sound; and Audiority’s Deleight was similarly delightful — a warm and flexible multi-tap delay. The resulting vocal was detailed and present, but I couldn’t get as ‘produced’ a sound as I did with Neutron. That said, the rest of the Scarlett’s Softube bundle scores you a multi-band mastering compressor (tasty) and a Tube Delay as well. FOOT IN THE DAW
The three interfaces provide different DAW options. Both the KA6 and 2i2 come with Live
Stacked up and ready to record, these three interfaces all deliver a great experience for a first-time user
Lite, which will get you started but you’ll hit your head on the eight-track ceiling pretty quickly, making it more of a marketing taster than a serious DAW option. The 2i2 also throws in Pro Tools First. It’s a free download but remains a capable home studio recorder, with mixing plugins, and a maximum of 16 tracks. If compromise isn’t your thing, PreSonus includes its full-fat StudioOne Artist Edition. This software alone is more than capable of producing real music, even an LP or two, and with the rest of software package, it makes for a compelling deal for bands or solo artists looking to record. PRODUCER BRO
For the producers and budding banger bros, all three packages include EDM and beatmakerfriendly content. PreSonus’ R&B sample VST is sadly outdated and unattractive, though its virtual
Stacked up and ready to record, these three interfaces all deliver a great experience for a firsttime user. All are plug ’n’ play, and play nice with a recent Mac. The KA6 and Studio 24C share Input/ Host crossfade knobs for direct monitoring, which allowed me to dial in a good mix of click and mic into my ears for a quick vocal take (or five). The Scarlett, however, offers a perfectly acceptable balance with no such tweaking available, only a direct monitoring on/off switch. Knowing the heritage of the Presonus XMax preamps on the Studio 24C, its time for the ultimate comparison. Who will win: Survivor or Contender? Old Faithful, or Hipster Box? The truth is, I’d be splitting hairs; in my test there was practically no discernible difference. If anything, the Focusrite pres had a little extra HF detail, thanks to their new ‘Air’ control. The minute differences were probably more mic technique etc. so in reality, the audio performance of your preamps will not be a deal breaker or maker in you choice of interface. BUNDLES OF JOY
All boxes come with comprehensive software bundles — a significant part of the value proposition when looking at an interface in this segment. I put them all to the test on my voice, to determine which would get me the best result. In the PreSonus corner: iZotope Neutron was first up, which has a reputation as a great sounding intelligent channel strip. Pulling up a Male Vocal
NATIVE INSTRUMENTS KOMPLETE AUDIO 6 PRICE $329 SOFTWARE PACKAGE VALUE Effects $170 Synths & Sounds $160 DAWs Free Total $330
PROS Great look & build, modular capability, 2 ’phones outs CONS Live Lite a bit meagre
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PRESONUS STUDIO 24C PRICE $249
hip-hop drummer from uJam sounds great and the Arturia Analog Lab Lite is serviceable, if limited. In contrast, it’s a close call between the KA6 and 2i2 for the dancefloor crown. They often go blow for blow, both offering months of access to subscription-based sound and loop libraries (sounds.com and Splice Sounds respectively), and analogue monosynths in Monark and BassStation. Both the KA6 and 2i2 offer heaps of downloadable content, but NI perhaps provides the knockout blow with Komplete Start, serving up more than you’re getting from Focusrite’s sample libraries. That said, NI doesn’t offer lifetime membership to any of its software. Focusrite’s big draw for established producers and engineers is Plug-In Collective, which regularly (every couple of months) offers members free plug-ins and vouchers from a new partner software company. Plug-In Collective is an enormous value-add: the software tools are always of a high quality, great sounding and useful, and you can score yourself some cracker deals. Meanwhile, KA6’s $40 NI store credit feels a little incipid in comparison.
SOFTWARE PACKAGE VALUE Effects $711 Synths & Sounds $110 DAWs $150 Total $971
PROS Killer software value, quality effects CONS Lacklustre samples, ’phones on the back panel
The truth is, I’d be splitting hairs; in my test there was practically no discernible difference.
TIME TO SPLIT HAIRS
To differentiate the hardware in this mid-tier interface segment is tough — you may prefer the sleek good looks of KA6 and the fact you’re plumbed into the NI software paradigm might make the decision easy. It’s also a no-brainer for people producing in Ableton who need some I/O to hook up a guitar, pedal or a synth module. Similarly, KA6 will be a go-to for Ableton Live producers taking their work to the stage, acting as a hub for vocals, FOH and monitor outs. Focusrite’s Scarlett 2i2 is the most popular interface going, and its 3rd gen refresh won’t do it any harm. Hardware-wise, it lacks the KA6’s flexibility, but goes blow for blow in the software fight. Bring your own DAW, and this is a great interface offering the pick of the plugins in this battle. For me, the real surprise here is the Presonus offering. Its onboard software is of monstrous value, and stands out as offering a truly complete package for the budding muso in need of a DAW and the production tools to get live tracks into their computer. Ponying up to the Studio 68C would get you similar I/O to the KA6, and provide everything needed to record a small band or even to do some drum tracking. A guitar-toting singer/songwriter will benefit best from the Studio 24C, especially with some mates to record overdubs.
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FOCUSRITE SCARLETT 2I2 3RD GEN PRICE $289 SOFTWARE PACKAGE VALUE Effects $345+ Synths & Sounds $24 DAWs Free Total $369+
PROS Sexy beast, ‘Air’ preamp EQ, Plug-In Collective! CONS Lacks MIDI
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REVIEW
AUDIO-TECHNICA 3000 SERIES
Wireless Microphone System Audio-Technica’s fourth generation wireless microphone systems offer a selection of interchangeable capsules, good sound quality, ease of use, and a range of 100m.
NEED TO KNOW
Review: Mark Woods
PRICE Beltpack System: from $999 Handheld System: from $999 AE5400 Condenser System: $1399 Charging Station: from $399
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CONTACT TAG (02) 9519 0900 tag.com.au
PROS Sound quality OLED screens Classic interchangeable capsules Sold build
CONS None
SUMMARY An affordable, modular, reliable and good-sounding wireless microphone system that includes options from their established wired microphones, while providing stiff competition for the major brands.
When it comes to professional wireless microphone systems, the major manufacturers from the USA and Germany have dominated the market for some time. They have the advantage of brand recognition and wellestablished product lines, but there are challengers in this international battle for customers. Japan’s Audio-Technica is probably the next biggest by sales, and they’re keen competitors. Based on their range of classic capsules, they make some of the world’s great microphones for stage and studio. The Audio-Technica 3000 Series is a professional quality FM wireless system operating in the UHF spectrum with true diversity operation (i.e. two antennas, each with its own receiving circuit). The fourth generation of the series, reviewed here, offers several improvements over the earlier third generation models including new OLED screens on the transmitters and receivers, and more interchangeable capsules. Operating in the FG1 frequency band (650MHz to 693MHz), the tuning bandwidth has been widened to a class-leading 43MHz and is capable of operating 32 channels concurrently. Also new is the ability to pre-select a backup frequency that can be quickly accessed from the transmitter if sudden interference strikes. RECEIVER
The centre of the system is the ATW-R3210 receiver. A new white-on-black OLED screen dominates the front panel and the display and is much sharper and clearer than regular LED screens; it can be read in daylight, for a start. Current frequency, mute status and other general info is displayed automatically. Twisting the rotary encoder beside the screen one notch either way switches the display to performance mode with big meters for RF and AF levels. The twist-and-push rotary encoder is intuitive, and navigation is made easier with the inclusion of a physical Back button. The rear panel offers the usual balanced and unbalanced outputs. Less common on a wireless receiver is the ground-lift switch. The receiver receives its signal from either the ATW-T3202 handheld transmitter or the ATW-T3201 beltpack, both new designs and both powered by two AA rechargeable batteries – ideally recharged in the optional but convenient ATWCHG3 smart charging dock. Great for side-stage storage, it holds two of either the mics or the beltpacks. For bigger productions a networkable version (ATW-CHG3N) allows up to five docks to be linked from one power supply. HANDHELD TRANSMITTER
The ATW-T3202 handheld transmitter is a new design and its metal case feels good in the hand. It’s fatter and longer than a wired mic, but well balanced and feels secure wherever you grip it. The paddle/antenna on the bottom of the transmitter contains a sheltered and configurable Mute button that can be set to either mute, disable the RF, or toggle between the working frequency and a pre-selected backup frequency. It’s raised so you can easily find it by feel when
you need to use it, but it’s small and takes a deliberate action to press it down. It needs to be held down for a second before registering, so it’s unlikely to be accidentally engaged. The other controls, including the on/off switch, are accessed by unscrewing the lower part of the transmitter body and sliding it down to expose the battery pack and nudge buttons for frequency selection and IR sync. It’s always something of a compromise providing access to the controls of a handheld mic while avoiding the natural tendency of performers to press the wrong button. Some folks won’t like the off/on switch being hidden away and it could be awkward if someone forgets to turn it on and the performer has to do it on stage (or throw it at the tech who gave it to them), but I think this is a good system, on balance. The Mute button is easy — but not too easy — to engage, and the things you should set before or after the show are all hidden under the cover. Placed closer to the capsule than before, the OLED screen on the handheld transmitter is small but the display is super bright and easy to read. The screen lights up for a few seconds whenever the Mute button is pressed, displaying the active frequency and the battery level. Setting up has been made as easy as possible. Available frequencies are scanned and selected on the receiver; pressing a button syncs the receiver to the transmitter, then you’re good to go. Up to 10 pre-programmed scan groups can be selected, depending on how many transmitters are required. A threshold setting chooses between prioritising the number of channels or the stability of channels. CAPSULES
There’s a great selection of screw-on capsules for the ATW-T3202, all based on Audio-Technica’s existing wired models. The thread mount is now standard industry size so capsules from some other brands would fit if you wanted to try them, although I’d be inclined to stick with the AT capsules to avoid any compatibility issues and ensure the capsule is specifically matched to the preamp/electronics in the transmitter. I’ve had lots of Audio-Technica mics in regular use for some time and know them well. The regular wired ATM510 (dynamic) and ATM710 (condenser) mics are part of the Artist Series range (AT’s entry-level professional range), and they and their ATW-C510/7100 wireless equivalents are good honest mics that offer solid construction, flat frequency responses and an overall natural sound; all for a reasonable price. The AE capsules cost a little more but they’re the highest quality and the best choice for professional applications. Based on the AE6100/4100 wired dynamic microphones, the ATW-C6100/4100 wireless equivalents are bright, quite aggressive vocal mics with extended high frequencies and a fairly accurate off-axis response. The AE6100 has long been my first choice for live drum vocals; it’s crisp, the hypercardioid pattern is tight and the spill from the kit sounds noticeably better than other dynamics. They’ve been around for a while
The system seems rocksolid in use. Auto-squelch maximises the claimed 100m operating range so that’s unlikely to be an issue, and once the frequencies are synced you don’t have to think about it much – just enjoy the sound.
but they now seem to be gaining popularity as vocal mics, especially with IEM users. Even better for some acts, the AE5400/3300 large diaphragm condensers use the same capsules as Audio-Technica’s renowned AT4050 and AT4033 studio mics. The wireless ATW-C5400 version is deservedly the most expensive of the wireless capsules on offer. I regard my wired AE5400 as my highest quality handheld vocal mic and reserve it for fine music vocals… and Bluegrass; it’s my first choice as a spot mic for banjo, and works on all acoustic instruments. Smooth as silk with sparkling detail and big low-mids, I recommend it for anything except loud rock. In a direct comparison the wireless versions sound almost exactly the same as the wired versions, and possibly better depending on your console. In theory there’s inevitably some loss of fidelity going wireless compared to using a good cable, but I can’t notice any here and the preamplification in the transmitter is beautifully matched to the capsules. I had to concentrate to pick them in a blind listening test, but with the mics in hand I could hear a couple of differences. Firstly, speaking right on the grille the wireless versions had more attenuation of the murky low frequencies. Secondly, the wireless versions have less handling noise than the wired versions. Handling noise is not the wired AE6100’s best feature, but it is noticeably improved on the wireless version. The same goes for the other capsules, too; the handling noise is not just at a lower level, it also has a more subdued character. I suspect the bigger and deader wireless body helps. BELTPACK TRANSMITTER
The ATW-T3201 beltpack is small, nicely made, and has a secure belt-clip. It feels tough, too. These things can take a beating, especially in theatrical applications. In the sheltered space between the antenna and the screw-in connector lives the Mute
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In a direct comparison the wireless versions sound almost exactly the same as the wired versions... I had to concentrate to pick them in a blind listening test, but with the mics in hand I could hear a couple of differences
switch, with its raised tactile guide to help you feel it in the dark, and the function button that either turns the RF off or engages the backup frequency. Like the handheld transmitter, the on/off switch is hidden under the clip-down front cover along with the same set of nudge buttons. You still have to open the cover if it’s not turned on when it should be, but it’s going to be a lot quicker opening the cover on the beltpack than unscrewing the body of the handheld transmitter. The beltpack is controversial in its use of new cH-style connectors to attach lavalier or clip-on mics. The cH-style connectors are smaller, secure and pleasing to use, but won’t accept mics with cW-style connectors from previous AT wireless systems without using the AT-CWCH adaptor cable (available from your AT dealer). Mic options for the beltpack start with the AT831cH cardioid condenser lavalier mic: suitable for vocals and instruments, it’s a tiny little mic that sounds big. The P892cH MicroSet and BP893cH MicroEarset are different types of headworn omnidirectional condensers, available in either black or beige. Completing the line-up is the BP894cH MicroSet headworn cardioid condenser, again available in black or beige. IN USE
The system seems rock-solid in use. Auto-squelch maximises the claimed 100m operating range so that’s unlikely to be an issue, and once the frequencies are synced you don’t have to think about it much – just enjoy the sound. Battery life is eight hours or more with good alkaline or Ni-MH batteries. Sitting the transmitters in the AT 30
ATW-CHG3 charging dock kept the rechargeables charged, and the routine reminded me to turn them off after use. I did keep accidentally unscrewing the grill over the capsule when trying to change capsules, but they don’t get changed often in normal use. Punk icon Lydia Lunch played the Theatre Royal in Castlemaine as part of her angst-filled spoken-word tour, ‘Dust & Shadows’, and just happened to ask if I had a wireless mic. She wanted to appear at the back of the room and walk slowly through the audience, speaking in a ghostly voice. It was a dramatic start to the show. For a moment I wondered if the sound was going to feed back as she walked right in front of the PA speakers, but not a hint. The ATW-C6100 drew immediate praise from the artist, who was impressed by its response with the mic held at arm’s length. It sounded great, shiny and detailed, dripping with reverb. CONCLUSION
All Audio-Technica gear is well-made, and I’ve found their products to be completely reliable over time. I’d choose the new 3000 series over a comparable digital system because the sound quality is great, it can handle more channels, and I prefer the latency-free nature of UHF wireless transmission. Competitively priced for the features on offer, the fourth generation of the 3000 Series is aimed at professional applications where audio quality is the highest priority... and shouldn’t that be all the time? It will provide some healthy competition for the leading brands from Europe and the USA.
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Interface, Headphones, Mic & XLR Cable
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REVIEW
KORG MINILOGUE XD & XDM Polyphonic Analogue Synthesizers Korg has updated its popular Minilogue, adding new features including the Multi Engine oscillator and the Effects Engine from the Prologue series.
NEED TO KNOW
Review: Jason Hearn
PRICE RRP $1199, Expect to pay $999 CONTACT CMI Music & Audio: (03) 9315 2244 or www.cmi.com.au
AT 32
PROS Same reprogrammable Multi Engine oscillator of the Prologue for a much lower price. Reprogrammable effects engine. Wider than regular analogue sound palette. Keyboard and desktop modules available.
CONS Limited modulation matrix. 16 Multi Engine slots may not be enough for some users as the range of third-party oscillators increases!
SUMMARY The xd packs a punch in terms of features, size and price. It’s a substantial refresh of the original concept, to the point that it is essentially a new and different synth.
In 2016 Korg released the Minilogue, their first polyphonic analogue synth since the ‘80s. This compact four-voice synth won popularity thanks to sounding superb and being inexpensive. Two years later, Korg followed with their flagship analogue polysynths, the Prologue 8 and Prologue 16 with eight and 16 voices respectively. The Prologues introduced the concept of a ‘Multi Engine’: a third programmable digital oscillator open for third-party development, and they also had an Effects engine open for third-party development. The Minilogue xd harnesses the versatility of the Prologue’s Multi Engine and Effects engine, yet maintains the Minilogue form factor at a price that’s not too much more than the original. The xd has inherited some of the cosmetics of the Prologue, with black knobs and a black faceplate, but maintains the unique curved form factor of the original Minilogue. As with its predecessor, the build quality is exceptional; the knobs are chassis-mounted and feel solid, providing high-resolution control of parameters where appropriate. The OLED oscilloscope display, a hallmark of the original Minilogue, is now larger. The quirky diagonal performance lever from the original Minilogue has thankfully been retired and in its place is a dual axis joystick located just above the keyboard. A series of 16 buttons flank the centre of the panel providing hands-on interaction with the step sequencer and much-improved access to the sub pages of the menu system. Finally, the audio output of the Minilogue xd is now stereo instead of mono. Korg has also released a desktop module version (Minilogue xdm) that will appeal to those looking for a more compact unit to take on tour. Korg added a new MIDI Poly Chain feature in a recent firmware update, providing an effective means of expanding the voice count by connecting additional units together via MIDI. Straight out of the gate, Korg has made the Minilogue xd Sound Librarian application available (for MacOS and Windows) providing patch library management and the means for uploading thirdparty content for the Multi Engine and Effects. The Voice Mode feature has been streamlined for the xd with its controls more intuitively located adjacent to the Oscillator section. The Poly, Unison, Chord and Arpeggio modes remain from the original. The Duo mode no longer has a dedicated mode, instead being integrated into Poly mode (in the latter range of the Voice Depth control). Also missing are the Note Delay (akin to a MIDI Note Chase effect) and the Sidechain modes. OSCILLATOR SECTION
The dual VCOs provide pure analogue waveforms (Saw, Triangle and Pulse) modulated by a Shape parameter which is routed to the LFO. For the Saw and Triangle waves the Shape parameter invokes wavefolding, gradually increasing harmonic complexity. For the Pulse wave the Shape control tweaks the duty cycle, and modulating it via the LFO yields PWM. VCO1 and VCO2 both feature Ring Modulation, Sync
and variable Cross Modulation. For the xd, these VCOs are complemented by a digital Multi Engine oscillator identical to that of the Prologue series. It offers three modes: User mode allows one to import 16 third-party oscillators; VPM mode provides a wavetable synth with 16 wavetables plus a noise generator; and ‘Noise’ mode offers four types of noise. For all Multi Engine types, a simple Shape parameter is offered for front panel tweaking. An additional six parameters are found in the Program Edit menu. The User mode is of the greatest interest. The first of its 16 slots is preloaded with Waves by Korg. At the time of reviewing the Prologue, third-party Multi Engine oscillators were yet to emerge. Since then, a small but growing core of developers has stepped up to the plate. Thankfully oscillators developed for the Prologue will import to the Minilogue xd. [Refer to ‘Third-Party Multi Engines & Effects’ for links.] For the sake of the review, I installed the Freeware oscillators offered by Peter Allwin and Jason Long. Exploration of these oscillators immediately showed how much the Multi Engine expands the sound palette, with many being rich in harmonic content. These additional oscillators inspired many new patches, and in several cases I even found it unnecessary to have the analogue VCOs layered to get satisfying tones with plenty of movement. The highlight for me was Jason Long’s developments; in particular, his Chips 2.0 oscillator which brings vintage digital game console sounds into play. He has been quite prolific of late, having released an additional two effect algorithms, Bucket and Hera, and also a preview video of a custom wavetable/sample import tool he has in development with full key-range mapping and more: https://tinyurl.com/yxbw3h6k With the sheer number of new oscillators I’ve discovered so far, it was apparent that having 16 slots for third-party oscillators may soon become a limitation. It’s important to note that swapping algorithms in and out of the synth will break patches unless the same oscillator types are loaded back into the correct slot number, because patches reference oscillators based on the slot they’re loaded into rather than by name. FILTER SECTION
The VCF is a two-pole (12dB/octave) low pass filter making use of Operational Transconductance Amplifiers (OTA). It has a rich, powerful and consistent sound when swept across the audible frequency spectrum, and is capable of selfoscillation. The cutoff frequency can be modulated in a bipolar manner from the EG or LFO. New for the xd (and just like the Prologue), the filter now features a drive stage offering two preset amounts of drive: 50% and 100%. VCF keyboard tracking can be engaged at 50% or 100%. With resonance at self-oscillation, you’ll find the key tracking is scaled to be in tune (great for high resonance white-noise leads). The four-pole mode of the original Minilogue is absent for the xd, fueling speculation that Korg has
repurposed the filter circuit from the Prologue for the Minilogue xd. Notwithstanding, to my ears it sounds slightly different. MODULATION MATRIX/SOURCES
The modulation matrix is basic, nevertheless the essentials are there to create ample movement. The xd has two envelopes: a traditional ADSR (hardwired to the VCA), and a simple AD routed to either VCF Cutoff or the pitches of VCO1 or VCO2. A single LFO is provided capable targetting one of either VCO Pitch, VCO Shape or VCF Cutoff at a time. There are additional LFO destination options in the menu, where it can target specific oscillators. (The multi-engine shape parameter has its own dedicated envelope also hidden in a menu.) On the back panel, you’ll find dual CV inputs designed to interface with Eurorack hardware. For those seeking more expansive modulation options than those available internally, you can easily integrate all manner of exotic modulation sources from your Eurorack modular synth. There is a total of 29 different modulation targets available for the CV1 and CV2 inputs. Additionally, the upward and downward travel range of the Joystick’s Y-axis may be independently mapped to the same list of 29 modulation targets.
...after exploring what the Multi Engine oscillator adds it’s clear that the Minilogue XD is capable of a sound palette far beyond a dual VCO synth.
EXPANDED (& EXPANDABLE) EFFECTS
The original Minilogue’s effect engine was a mono bucket-brigade analogue delay that was a touch primitive and noisy. The effects section has been vastly overhauled for the xd, inheriting the same lush-sounding stereo DSP Effects Engine of the Prologue series. The first slot provides Modulation effects: Phaser, Flanger, Chorus, and Ensemble. The second and third slots offer Reverb and Delay respectively. Only the Rate and Depth parameters are available to tweak – all other parameters are hardwired in various factory presets. While each algorithm sounds excellent and the presets are well-appointed, like anything preset, I did crave the ability to tweak further. Similar to the Multi Engine oscillator, additional effect algorithms can be installed with 16 slots for AT 33
I found myself inspired to create new patches every time I had a session with it.
the Modulation FX and eight slots each for Reverb/ Delay effects. At this point, I can’t help but recall the excitement of loading third-party effects into my Ensoniq EPS16+ and ASR10 samplers back in the ‘90s. A brief run-down on what is available is covered in ‘Third-Party Multi Engines & Effects’. ARPEGGIATOR & SEQUENCER
Like its predecessor, the Arpeggiator is implemented as a feature within the Voice Modes, rather than being an independent entity, and now features a Latch function. Thirteen arpeggiator modes are available on the Voice Depth control with some even providing polyphonic modes, which is a nice touch. Rather than a dedicated octave span setting, you’re limited to what is predetermined within these modes. The decision to implement the Arpeggiator within the Voice Modes means it’s not possible to have arpeggiated patches with the unison engaged simultaneously. A polyphonic Sequence can be attached to each patch memory independently and may have up to 16 steps. Sequences can easily be entered in step mode or recorded and overdubbed in real-time alongside a metronome. A powerful Motion sequence feature allows capture of knob movements for up to four parameters simply by recording movements during playback. It’s also possible to have Motion sequences defined without having any note data in the sequencer, effectively providing four beat-synced, 16-stage looping envelopes – a useful means to extend the modulation facilities of the synth. Because the VCO levels can be automated, it’s possible to simulate complex multitimbral sequences by bringing different VCOs and/or the Multi Engine in and out of the mix on different steps. This is further enhanced by the fact the Multi Engine can be set to bypass the VCF within the menus. The possibilities are capably demonstrated in a video tutorial by BoBeats, here: https://tinyurl. com/y3lgnx55. My only real disappointment is that Sequences can’t be transposed in real-time by striking notes on AT 34
the keyboard as part of a live performance; which is a missed opportunity because that’s the essence of making step sequences really come to life. I had issues with the Sync Source setting in the Global Edit menu providing only ‘Auto’-based settings for slaving to external clocks, either via MIDI or USB. For example, if you are playing the Arpeggiator (Latched) whilst slaved to MIDI clock and you stop the transport, the unit assumes you no longer want to use an external clock and, after a brief moment, resumes playback. Hopefully, Korg considers adding non-auto sync options. CONCLUSION
The Minilogue xd packs a punch in terms of features, considering its size and price. Far from being a mere incremental feature bump, the Minilogue xd is a substantial refresh of the original concept – so much so that it really is a different synth: sounds created on the original Minilogue cannot be imported into the Minilogue xd because the architecture differs too much. Despite its limited modulation options, after exploring what the Multi Engine oscillator adds it’s clear that the Minilogue xd is capable of a sound palette far beyond a dual VCO synth. Once you program your own patches to taste and think inventively to achieve movement within sounds, it really sings. The icing on the cake is the inclusion of the Prologue’s Effects engine; on the original Minilogue it was difficult to evoke really big lush sounds without using an external effect unit, but with a few knob twists and switch flips in the Effects section of the Minilogue xd you’ll quickly find yourself in sonic nirvana. It was effortless to coax everything from angular aggressive digital basses and snarling dirty leads through to floaty sublime keys and pads. I found myself inspired to create new patches every time I had a session with it. If you’re in the market for a modern four-voice polyphonic analogue synth with the extra fruit that only hybrid digital synths can provide, the Minilogue xd is definitely worth checking out.
THIRD-PARTY MULTI ENGINES & EFFECTS The Multi Engine has been out in the wild since the Prologues hit the market, and several developers have been getting busy. With the Minilogue xd bringing the Multi Engine to the masses in a much lower priced synth, I expect the offerings will continue to grow. Sinevibes https://www.sinevibes.com/korg/ Multi Engine Oscillators: KorgBent, KorgTube, KorgTurbo. Tim Shoebridgehttps://sellfy.com/ soundmangling/ Multi Engine Oscillators: String, Pluck. DirtBoxSynth https://www.dirtboxsynth.com/ Multi Engine Oscillators: Complete Oscillator Collection (six different oscillator types) Delay Effects: Exotic Delay pack (seven different advanced delay effects) Modulation Effects: Filter pack, AMPit. Peter Allwin https://github.com/peterall/ eurorack-prologue Multi Engine Oscillators: 12 derivatives of Mutable Instruments’ Plaits and Elements (Freeware). Jason Long https://github.com/hammondeggs Multi Engine Oscillators: Chips, Souper, Percy (Freeware) Delay effects: Arrhh, Bucket (Freeware) Modulation effects: Hera (Freeware)
AT 35
REVIEW
ANTELOPE ORION 32+ GEN 3 MADI & USB 3.0 Interface The Orion 32+ Gen 3 is the latest evolution of Antelope’s Orion series of interfaces. Benjamin Lancaster put it to the test...
NEED TO KNOW
Review: Benjamin Lancaster
PRICE $5499 CONTACT www.antelopeaudio.com
AT 36
PROS Pristine sound quality and rock-solid stability Routing is comprehensive and relatively easy
CONS No internal monitoring system Requires D-Sub looms
SUMMARY The Orion 32+ Gen 3 provides a perfect bridge between a console and DAW, and is also very well suited to live recording applications. It’s a considerable investment, but does not disappoint.
Let’s just begin by saying that, as a long-time Apogee devotee, I had some serious soul searching to do after a fortnight with the Orion 32+ Gen 3 audio interface. This is a professional interface for those in search of pristine conversion and hyper-flexible integration. It offers 32 analogue line level inputs, 32 analogue line level outputs, a stereo analogue monitor output and a range of digital I/Os including MADI, ADAT and SPDIF. There are connections for wordclock in and out, and Thunderbolt and USB-A ports for interfacing to your computer. Antelope’s marketing materials claim exemplary A/D and D/A conversion, flawless clocking thanks to their proprietary 64-bit Acoustically Focused Clocking (AFC), and superlow ‘sub-millisecond’ latency. I was keen to test the Orion in my ‘Analogue DAW’ recording rig (see AT133). The setting up process was a fairly typical affair for anyone who has integrated hardware into their DAW. Online registration and downloading all the affiliated software was par for the course, and I was up and running in a little under 30 minutes. INPUTS AND OUTPUTS
Marrying the Orion with my current setup was fairly seamless. The eye-watering array of I/O options have pretty much every engineer’s connectivity preferences covered and, through a miracle of engineering, the comprehensivelyequipped back panel is so neat and tidy you could set your watch to it. The stereo Monitor output uses a pair of TRS sockets, making it easy to connect to. The rest of the Orion’s analogue inputs and outputs are accessed via D-Sub connectors conforming to the eightchannel AES59/Tascam wiring format. These, along with the Orion’s broad range of digital options, means it would be difficult to find a piece of gear you can’t connect to it – assuming you have the requisite cabling on hand! The Orion does not come supplied with the D-Sub breakout cables needed to actually use it, although to be fair Antelope cannot anticipate how many looms, what lengths and what type of connectors each user would require on the end of those looms; some would want XLRs, others TRS, and even others might want to solder it directly to a patch bay. Just be warned that before you can plug it in you’re going to be in the red for at least one D-Sub cable of some design or another. This is not plug-and-play technology!
For this review I opted to go straight from the outputs of my Analogue DAW’s preamps into the Orion’s D-Sub analogue inputs. While making all of those connections I found myself daydreaming about how brilliantly the Orion would handle multitrack recordings of live shows by connecting to the digital output of a live console via a single MADI connection. With its stable build and near limitless routing abilities, it would be more than comfortable in this situation.
The Orion 32+ Gen 3 provides a perfect bridge between a console and DAW, with a near-infinite amount of routing possibilities and enough channels to handle most large format consoles... It’s also very well suited to live recording applications.
SOFTWARE
At first glance the Orion’s software panel is sleek, modern, well designed and, most importantly, scalable! There is nothing worse than a minuscule panel on a 27-inch screen that you cannot see, let alone operate; so, thank you Antelope! Tabbing through the different sections while making selections is seamless and intuitive. The Mixer is laid out over five windows to represent the seemingly endless channels, and cycling through them is a breeze. The wheels began to come off the cart for me in the Matrix section. Let me be very clear that this is an evaluation from an ordinary person trying to make sense of an extraordinary routing system, and the fault here does not lay entirely with Antelope. The Matrix section is, in a word,
brilliant; however, some of the labelling decisions are questionable. Though the Matrix is colourcoded and it is easy enough to assign I/O, I got completely confused trying to assign outputs to my monitoring section. The loose cog in the machine is in the naming of the I/Os. ‘TB Play 1-32’? If you are like me and did not initially watch the videos or read the instructions, ‘TB Play 1-32’ could be anything! What I later discovered is that it’s the assignment for ‘Thunder Bolt Playback Channels 1-32’ – the digital returns from your DAW via Thunderbolt. Once this got cleared up, the ability to route DAW outs to the Orion’s analogue outputs became blatantly obvious and actually quite easy. Maybe Antelope should re-label it as ‘DAW Returns 1-32’ or similar for those of us who are keen to get started without watching videos or reading instructions! RECORDING
Low latency is one of the big talking points of the Orion series, and for good reason. The ‘submillisecond’ (ie. less than one thousandth of a second) latency is not long enough to throw off any musician, and I would call out those who say they are affected by it. For me and the artists I worked with – and this included Midi-based musicians – the Orion performed outstandingly. With an electronic drum kit triggering Kontakt’s ‘Abbey Road’ series of drum kit plug-ins, the response and feel of the audio coming back through the monitoring section was smooth and felt to be in perfect time. Recording audio was obviously no drama due to the vast number of channels; the only limit was how many preamps I was able to stick in front of it! The Orion’s software comes provided with a number of DSP plug-ins from their FPGA FX platform, including their Master De-Esser, Power Ex expander, Power Gate, Power FFC compressor, Clear Q five-band parametric EQ, and Aura Verb reverb. Additional plug-ins can be purchased and added later, with over 30 to choose from. I must admit that I only gave the installed plug-ins a cursory glance because I already have an exhaustive library of top-quality plug-ins, and I assume anyone in the market for the Orion would have their own plug-ins as well. Having said that, the Orion’s built-in reverb in the monitoring section is a stroke of genius. The vocalists I recorded with the Orion were so grateful and happy to have reverb AT 37
quickly added to their headphone mix without the time-sucking task of patching in a unit or dealing with troublesome DAW latency issues that often cause unnatural artefacts. By quickly dialling in a bit of reverb, I kept the workflow uninterrupted and ultimately more fruitful than it would have been otherwise. The reverb has a comprehensive feature set, sounds pretty good and serves its purpose brilliantly. SOUND
I am not a fancy man when it comes to the monitors I use for mixing records. I have a pair of Event 20/20s that I’ve made many records with; I know them well and I trust them. After installing the Orion I pulled up a current track that I’d been previously mixing through my Apogee converters. The sound through the Events came alive in a way I had not previously heard. The soundstage widened, there was greater separation between instruments, and the plane of sound felt as if it had jumped towards me by another six inches or so. The instruments that were compressed sat back in the way you would expect while the more dynamic elements leapt from the speakers, creating a thoroughly pleasing three-dimensional space. These improvements in soundstaging and separation were even more evident when I switched to my AKG K712 open-back headphones. In addition to being an insufferable nerd about recording and mixing, I lead a double life as an insufferable audiophile. Perhaps the two are not mutually exclusive? After the Orion’s success in the studio, I decided to take it into my living room, connect it to my hifi and see how it faired in the hifi context of listening for enjoyment rather than making critical mix decisions. I have an extensive range of hifi equipment but my main system is AT 38
built around a Cambridge Audio DacMagic Plus driving a pair of B&W 606 speakers through a Harman Kardon HK980 amplifier. I connected the Orion’s Monitor outputs to the HK980 with a pair of 6.35mm jack to RCA leads. I don’t think it will shock anyone who has heard Antelope’s conversion to find out that, yet again, the sound quality was peerless. The same soundstaging mentioned above came into sharp relief as the speakers breathed out Nick Cave and the Bad Seeds. It was an enveloping aural experience, but buying the Orion 32+ as a stereo hifi DAC would be a ridiculous notion to most rational audiophiles. It was fun though, and it certainly showed the capabilities of what the Orion can do. I fear my Apogee days may be limited… GRIPES
I have a couple of gripes about the Orion 32+ Gen 3, and they are to do with its monitoring section and front panel. Whilst I love the idea of having routing presets, access to these from the front panel seems superfluous. I would’ve preferred a dedicated Monitor volume pot so I did not have to rely on software to control the level to my main speakers, which is not ideal and hinders workflow. It’s not a problem if you have a dedicated monitor controller, but I would prefer a front panel volume control for this purpose. A headphone socket on the front panel would also be desirable. At only 1RU high the Orion has the potential to be a semi-portable device but offers no way of independently hearing anything coming out of it without, at the very least, connecting a headphone amplifier. I know this is nitpicking and the Orion is not marketed as a stand-alone interface, but having buttons to select routing presets seems like a waste of precious front panel real-estate when a dedicated Monitor out volume
control and headphone socket would be far more useful. If Antelope is reading this, please consider these suggestions for Gen 4. WRAP UP
The Orion 32+ Gen 3 provides a perfect bridge between a console and DAW, with a near-infinite amount of routing possibilities and enough channels to handle most large format consoles. It’s poised to compete with top level studio-grade interfaces like Avid’s HD Systems, albeit with more of a ‘free market’ mindset. It’s also very well suited to live recording applications – nothing could be simpler than a MADI connection into the Orion to create a bridge from a live console to a DAWbased recording system. Those with home studios would consider it overkill for their needs; for that market Antelope offers the Zen Tour and Orion Studio HD, and they make this distinction obvious through their marketing. At $5499, the Orion 32+ Gen 3 is a considerable investment but if you’re looking for an ingeniouslydesigned and spectacular-sounding alternative to the status quo of studio grade interfaces, you will be sated and inspired.
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antelopeaudio.com AT 39
REGULARS
LAST WORD Meaningful Comparisons Column: Greg Simmons
In this issue I reviewed Røde’s TF-5 small diaphragm cardioid condenser microphone and made comparison recordings to share with readers. I put the TF-5 against two established favourites I’m familiar with: Neumann’s KM184 and DPA’s 4011. Between the three, I aimed to create a set of comparison recordings that allowed the listener to determine where the TF-5 sat on the price/performance scale. The process of making meaningful comparison recordings was harder than it seemed, with each method inviting uncertainties that rendered the comparisons meaningless. RAISING THE BAR
For my review purposes the recordings only needed to highlight enough differences between the mics so that I could make informed opinions, but those opinions were also informed by a lot of background stuff: I made the recordings and knew what the instrument sounded like in the room, I’d experimented with different microphone and instrument positions, asked the musician to play specific things to highlight differences, exchanged opinions with others at the session, and even went back and did it again to confirm my initial impressions. All that background context is lost when the recordings are passed on to someone who wasn’t at either of the sessions. So if I was going to make comparison recordings for readers to form their own opinions, those recordings needed to be meaningful on their own. The standards of the recording quality, and the musician’s playing, suddenly got higher. TRIANGULATION
How hard can it be to compare three microphones? Stick them side-by-side, match the gains and hit record, right? That’s okay if you’re miking from a distance, where a few centimetres between diaphragms will probably be irrelevant. But it’s not okay for a close-miking test. For the comparison recordings, I used acoustic guitar, and anyone who has miked an acoustic guitar from 30cm or less knows that moving the microphone a few centimetres can make a noticeable difference to the tonality of the captured sound. It’s the kind of difference that can render the comparisons meaningless: are you comparing microphones or miking positions? So the goal was to get the diaphragms as close to each other as possible. But how? AT 40
ONE MIC AT A TIME
Ideally all three diaphragms would be in exactly the same spot. That’s not physically possible without doing three separate takes, one microphone at a time, and that’s only going to provide meaningful results if the performer can give exactly the same performance from exactly the same position, every single time. Separate takes means separate performances, and that means it’s impossible to tell if tonal differences between recordings are due to the microphones or the performance. The comparison recordings are no longer meaningful. THREE MICS AT ONCE
How about laying two of the microphones sideby-side, laying the third along the top of them like stacking logs, and taping them together? They all have tubular bodies of similar diameters, and when bundled this way each diaphragm would be as close as possible to each other diaphragm. There are two problems with this approach. First, these are all end-address cardioids with ports behind the diaphragms. Bundling them together as described here means each microphone will have its ports obscured at two points (one point from each of the other microphones). Whether or not that will affect the tonality of any of the microphones is debatable, but as long as it is debatable the comparison recordings are meaningless. Second, each of the three microphones has an outer diameter of around 20mm, so the closest any two diaphragms can get to each other is around 20mm centre to centre. That distance is unavoidable, of course, but could result in each microphone capturing a different tonality from the instrument if the relative orientation of the microphones is not considered. Which orientation provides the least tonal differences on the instrument being reviewed: vertical, horizontal or diagonal? Get it wrong and the comparison recordings become meaningless. UNMATCHED PAIRS
I decided to focus on two microphones at a time: the TF-5 and the one I was comparing it against. I’m not comparing the KM184 against the 4011, so I don’t need them in the same test. The next decision was placement. As any experienced engineer knows, the best spot for one microphone may not be the best spot for another. To avoid playing favourites, I used a consistent
placement of 30cm from the guitar, perpendicular to the soundboard, with both microphones aimed at the point where the guitar’s neck joins the body – my preferred starting point when recording acoustic guitar. For the distant miking tests I pulled the microphones back while keeping them focused on the same spot. For orientation, putting one microphone above the other (rather than side by side) yielded better tonal consistency during subtle performance movements of the guitar – the tonal differences between microphone positions was significantly smaller than the tonal differences between the microphones themselves, and therefore irrelevant. I left a small gap between the microphones to avoid any possibility that one microphone was blocking the ports of the other. To ensure consistency between comparisons, the guitarist composed four short pieces to reveal different aspects of each microphone’s performance: long slow strums, fingerpicking, muted strums and full-bodied strums. Finally, I sent a blend of the two microphone signals to the headphones so the guitarist could ‘work’ the microphones; staying on-axis and subtly moving closer and further back as desired, just like any normal recording session. MATCHING PERCEIVED VOLUMES
The final step in preparing the comparison files was to carefully match the perceived volumes of each pair of recordings. This was essential because the TF-5’s frequency response includes a shallow dip from 2kHz to 7kHz, falling to almost -1dB from 4kHz to 5kHz. This dip includes the range of frequencies that human hearing is most sensitive to, meaning the TF-5 will probably sound lower in perceived volume when compared to a microphone without such a dip. In contrast, the KM184 and 4011 both have a subtle boost in the upper part of that same range of frequencies. Any attempt at simply matching the peak levels or matching the waveforms consistently resulted in the TF-5 sounding dull and uninteresting in comparison to the others when, in fact, it was being reproduced at a slightly lower perceived volume. As any experienced engineer or hi-fi salesperson knows, if you want someone to prefer one sound over another make it louder. Without carefully matching the perceived volumes, the comparison recordings would always favour the louder of the two files and would therefore be meaningless.
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