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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 Greg Simmons Mark Woods Ewan McDonald Guy Harrison
Graphic Designer Daniel Howard daniel@alchemedia.com.au Advertising Philip Spencer philip@alchemedia.com.au Accounts Jaedd Asthana jaedd@alchemedia.com.au Subscriptions Sophie Spencer subscriptions@alchemedia.com.au Proofreading Andrew Bencina
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. 01/07/2020.
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COME IN & CHECK OUT OUR NEW SHOWROOM
Turramurra Music is Sydney’s leading professional audio equipment destination. Our recently renovated live sound PA showroom features a wide range of mixing consoles, from compact analogue and digital rack mixers, through to large format digital consoles from leading manufacturers including Allen & Heath, Midas, PreSonus, Yamaha and more. Plus, we have just been appointed as one of Australia’s only official DigiCo dealers. We have Sydney’s largest range of consoles available for hands on demo in store, along with 40+ individual PA speakers – and that’s just our live sound area.
Our dedicated PA specialist staff can support you with any enquiry large or small, from upgrading your mixer, speakers or mics to complete system design and installation. We also provide mixer training for individuals and groups, and have dedicated education and church sales staff catering specifically to the individual needs of these customer groups. Come in and see us in store, or contact us today and discover why after over 40 years in business we have become Australia’s leading musical instrument and professional audio destination.
1267 Pacific Hwy, Turramurra, NSW AT 4
DiGiCo | SD9 Boasting 96 channels at 48kHz/96kHz, the DiGiCo SD9 delivers ultra-high performance digital mixing in a surprisingly compact package. Stealth Digital Processing and floating point Super FPGA technology combine with 24 touch-sensitive motorised faders, quick access function buttons, dedicated multi-function knobs, and a 15-inch, high resolution touchscreen to deliver renowned speed with an efficient and familiar workflow. This makes the SD9 a formidable option for high end touring, broadcast, theatre and house of worship applications. New to Turramurra Music, come in and demo the DiGiCo SD9 in store now!
Aston | Stealth Featuring 4 voice settings Vocal 1, Vocal 2, Guitar and Dark, the Aston Stealth offers a range of world class sounds to suit almost every application. The Stealth is a broadcast quality microphone perfect for podcasters and budding studio musicians. It features a proprietary internal shock mount, excellent side rejection and can be used with or without phantom power. An auto-detect circuit senses 48 V phantom power and engages an active built-in class-A preamp with 50 dB of boost (an industry first). Come in for a test drive in our studio showroom today!
Yamaha | MODX6 Designed for musicians on the go, this lightweight production synthesizer features playback of a huge variety of sounds from Pianos, Drums, Bass, Guitars, Brass and Woodwind and more. It also features an 8-operator, fully controllable FM-X engine, seamless sound switching for smooth performances with no cutoff in envelopes or effects. The MODX allows users to bring in their own custom samples or from Synth Libraries directly from Yamaha. Come in-store and test drive the MODX in our Synthesizer Showroom today!
Sennheiser | PRO In Ear Monitors – IE400 / IE500 The new Sennheiser PRO in ear monitors offer a natural, detailed and incredibly accurate monitoring experience. Featuring a single premium broadband transducer, the IE400 and IE500 are able to produce natural sounds free from crossover effects and phasing issues – a drawback often overlooked in multi-driver IEMs. The IE400s offer excellent monitoring for backline musicians, with a punchy response and pronounced bass frequencies. The IE500s offer a beautiful monitoring experience for vocalists and acoustic guitarists with exceptional midrange and treble frequency reproduction and a wide soundscape allowing for excellent stereo separation. We have a demo sets of both models in store, so contact us to arrange a demo.
Visit www.turramusic.com.au • Call (02) 9449 8487 • Email hitech_sales@turramusic.com.au AT 5
COVER STORY
16
i For Detail: Recording Bon Iver
ISSUE 65 CONTENTS
14
Feeling Good: Michael Bublé Live
RØDE VideoMic NTG Shotgun Mic
12
Universal Audio Apollo Twin X Audio Interface
Dynaudio Core7 Active Monitors AT 6
Mixing Synth Pop (Part III): Synths
10
42
Korg Volca Nubass Vacuum Tube Synth
Meyer Sound X40 Ultra Active Loudspeaker
30
40
28 36
INDUSTRYLEADING CONVERTERS
FAST & RELIABLE CONNECTIVITY
LATENCY-FREE MONITORING WITH EFFECTS
$
269.99 RRP
$
499.99 RRP
2 x 2 USB 3.0 Audio Interface
6 x 4 USB 3.0 Audio Interface
1,199 RRP
$ 16 x 16 USB 3.0 Audio Interface RECORDING PACK
$
449.99 RRP
Interface, Headphones, Mic & XLR Cable
steinberg.net YA M A H A MUSIC AUST R A L I A PROUDLY DIST RIBU T ES ST EINBERG PRODUC T S IN AUST R A L I A
yamahamu sicau Terms & Conditions: The prices set out in this advertisement are recommended retail prices (RRP) only and there is no obligation for Yamaha dealers to comply with this recommendation. *Errors and omissions excepted. AT 7
REVIEW
ARTURIA AUDIOFUSE 8PRE USB-C Audio Interface Review: Preshan John
AudioFuse was an outstanding maiden interface for Arturia. Wrapped in a futuristic design and several thoughtful features were squeezed into the desktop package. Arturia has expanded the AudioFuse’s appeal by developing a rackmount version with more I/O. Meet the AudioFuse 8Pre. Carrying over the innards from its predecessor, the most obvious difference of the new rack-able version is six extra preamps; eight in total, as opposed to two in the first and four in AudioFuse Studio. Metering has a similar look with the bright LED strip sitting next to each channel’s gain pot. Although the AudioFuse 8Pre only has one set of headphone outputs, we still get both 6.5mm and 3.5mm options — no stress if you can’t find that elusive adapter. IN ACTION
NEED TO KNOW
Firing up the AudioFuse 8Pre couldn’t be easier. Simply install Arturia AudioFuse Control Center, plug the interface into your computer and you’ll be auto-prompted through the usual protocol — firmware update, software updates etc. Arturia’s DiscretePro preamplification appears in all AudioFuse interfaces and features separate circuitry for mic and line signal paths. Preamp specs state a noise floor of -129dBu and a dynamic range of nearly 120dB. A +10dB boost button takes the preamp’s gain range to a hefty 72dB with
PRICE $1199 CONTACT CMI Music & Audio: (03) 9315 2244 or www.cmi.com.au AT 8
a -20dB pad also available. It’s a full, big sounding pre which stacks well in mixes, and the bucketload of gain is very quiet — dust off those ribbon mics! I love the tall and bright meters which make levelsetting a breeze. A switchable high-pass filter would have nicely finished it off. Audio resolution is up to 96k and the DAC will give a respectable 119dB of dynamic range on the monitor outputs. SOFTWARE PACKAGE
The AudioFuse Control Center software is fantastic. I love the ‘Add/Remove Tracks’ option to declutter the GUI by selecting only the inputs you’re using to appear in the monitor mixer section. Tabs at the top neatly show or hide the Input, Mixer and Output segments. Drop-down menus let you assign what’s feeding each set of outputs. ‘Intelligent’ inputs mean the toggle controls stay greyed out until you physically plug in a source. Another ingenious hack is track grouping. Using the buttons marked 1-4, you can swiftly allocate channels to colour-coded groups by clicking Edit, then a number, then the tracks you want in that group. In a recording situation this can really speed up your workflow, be it grouping drum tracks or choir mics for monitor mixes. Mutes and Solos are grouped too. You don’t find this option in many interface routing applications and I’m glad Arturia threw it in. My only criticism is I would have liked the ability
PROS Quality preamps with heaps of gain Routing software is intuitive and comprehensive
CONS Two headphone outs would be nice Software allows only one cue mix
to create more than one cue mix. An extra set of headphone outputs wouldn’t hurt either. TWO TRICK PONY
Another useful function of the AudioFuse 8Pre is that it can be a standalone eight-channel ADAT preamplifier. Push and hold the button on the far right corner to kick into ADAT mode, hook up an interface and you have instantly expanded your input capabilities. This is really cool if you own multiple interfaces. If you want to do what Arturia tells you, snatch two AudioFuse 8Pres and use one in each mode for a 16-preamp rig. Thrown into the package is the AudioFuse Creative Suite — a bundle of seven Arturia plug-ins including Analog Lab Lite, three preamp emulations, and recreations of a Moog ladder filter, 1176 compressor and Roland RE-201 Space Echo delay. In use you’ll find them to be high quality professional tools that, as Arturia promises, you will actually use. I am especially a fan of the lushsounding preamp trio. My overall experience with the AudioFuse 8Pre was very favourable. It just works: no glitches or speedbumps, it sounds great, software inclusions are usable, and Arturia’s attention to detail is evident in the unit’s ergonomics, aesthetics and functionality. Certainly the AudioFuse would be high on my wish list should I be chasing a new rackmount interface.
SUMMARY Life is too short to use troublesome interfaces. Be of good cheer – Arturia’s AudioFuse 8Pre is about as reliable as they come. The first rackable rendition of the AudioFuse concept, 8Pre functions on its own or as a standalone ADAT expander with clean pres and fantastic control software. Bundled with instrument and emulation plug-ins you’ll actually use, AudioFuse 8Pre is a workhorse interface that’ll just work.
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REVIEW
UNIVERSAL AUDIO APOLLO TWIN X Thunderbolt 3 Audio Interface with DSP UAD takes Apollo out of the rack and onto the desktop.
NEED TO KNOW
Review: Preshan John
PRICE $1699 CONTACT CMI Music & Audio: (03) 9315 2244 or www.cmi.com.au AT 10
PROS Excellent sounding interface with expandability Generous free bundle of DSPaccelerated plug-ins Desktop design great for home studios
CONS None
SUMMARY If you can’t work out of a million dollar studio, the UAD Apollo Twin X ‘studio in a box’ concept is the next best thing. With onboard DSP driving pristine plug-in emulations, this desktop interface features two Unison-enabled mic pres, monitoring control, and ADAT expandability. As a bonus, Apollo users get the UAD Luna recording system (Mac only) for free when released later this year.
A studio in a box. That’s the basic premise of a Universal Audio interface. With the DSP-enabled line of Apollo interfaces, UA has done a fantastic job recreating the signal chain workflow of an outboard-laden analogue studio within a digital microcosm. The UA Apollo Twin X is built on this idea, yet takes on a new outer shell. Of the many members in the Apollo family, the Twin X is the most compact. The sturdy desktop unit is less than a handspan across, totally backpack-able and rugged enough for an itinerant life. It’s also Thunderbolt 3 so latency reaches new lows when tracking through UA’s emulations of prestigious outboard models. UAD-2 Duo processing is the brains of the DSP, letting you load up the plugs without stressing your PC (Quad Core on the larger Apollo Twin x4). Two Unison-enabled preamps with alterable input impedances allow for more accurate preamp emulations from Neve, Helios, API, Manley, Universal Audio and more. Each offer 65dB gain. Conversion has been redesigned giving a commendable 127dB of dynamic range on the monitor outputs. UA has considered expandability as well. Unlike the smaller desktop Arrow interface, the Apollo Twin X not only has an ADAT input but it can be cascaded with up to four Apollo devices over Thunderbolt to increase your studio’s I/O and DSP potential.
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IN THE STUDIO
I am a fan of the Apollo Twin X’s ergonomics. The Preamp and Monitor buttons dictate the parameter controlled by the central knob while the row of buttons along the bottom toggle preamp functions such as pads, hi-pass filters, phantom power and mic/ line switching. Surrounding the main knob is a striking arc of led rectangles giving you a quick readout of what level you’re at — be it headphone or monitor outputs, preamp gain etc. It’s quick to figure out and easy to use. The built-in talkback mic is always valuable on desktop interfaces of this kind, as is the front facing headphone output and DI input. Perfect for bedroom or garage recording conditions. UAD Console is your virtual home where signal chains are built and the Apollo Twin X is controlled. The software downloads quickly onto a Mac and after a quick restart you’re good to go. I recommend memorising the four keyboard shortcuts (Cmd + 1/2/3/4) to toggle through important views across the input channel strips — Overviews, Inputs, Inserts and Sends. Placed on the right are the two Aux return and Talkback channels. Overall the GUI feels familiar and comfortable. VIRTUAL STUDIO
On to the good stuff – those juicy plug-ins! UAD has bundled the Apollo Twin X with the Realtime Analog Classics Bundle; a generous helping of emulation plug-ins. Guitarists get a Marshall Plexi emulation, an Ampeg SVT-VR for bassists, UA 610 preamp, LA-2A and 1176 LN compressors, a couple of Pultec EQs, plus a few effects. By default the entire library of UAD plug-ins sits alluringly in the menu but you can hide nonpurchased plugins in Console’s settings. All the emulations are stunning, as you’d expect from a company largely creating renditions of its own hardware (e.g. the 1176 and LA-2A). Putting your mics through UA’s classy virtual signal chains is a truly rewarding experience and your tracks will thank you for it. Whether it’s feather light LA-2A compression on a vocal or the Unison-enabled responsiveness of a DI’d electric guitar through a Marshall stack, the Apollo Twin X delivers a versatile and rich recording experience in a little package.
An Apogee Symphony Desktop Worth $2199 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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REVIEW
RØDE VIDEOMIC NTG Shotgun Microphone
Is it a mic? Is it an interface? Will it pair with your camera, smartphone, computer or tablet? Rode’s versatile VideoMic NTG says yes to all. Review: Preshan John
Rode’s VideoMic NTG sees two worlds collide — namely, the VideoMic family of utilitarian tools for videographers, and the NTG line of broadcast-style shotgun microphones aimed at audio professionals. The result is a versatile supercardioid self-powered videography mic up for grabs at a reasonable price. Rode has ensured VideoMic NTG isn’t picky about what it’ll work with. This is the only shotgun mic I can think of that will happily function with a smartphone, laptop, camera, tablet, field recorder, you name it. There’s more than meets the eye with this little mic. FUNCTION ROOM
NEED TO KNOW
Rode’s focus on functionality shows itself in a number of nifty little hacks. A little cable guide on the Rycote mount lets you stow excess cable length and avoid a large dangly loop around the camera. The gain pot is an ergonomic ring at the base of the microphone with a clear numeric readout as you turn it. The USB-C connector not only functions as a charging port but also allows the mic to interface with a computer as an input device e.g. to a DAW; perfect for a quick podcast recording or putting down a GarageBand scratch track of a musical
PRICE Expect to pay $369
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CONTACT Røde: (02) 9648 5855 or info@rode.com
idea. No drivers were necessary with my Mac — I plugged it in and saw audio instantly thanks to the auto-sensing power toggle. As the icing on the cake, the VideoMic NTG’s TRRS port turns into a headphone output when used this way. The first button atop the mic toggles between 75Hz/150Hz HPF rolloffs while a long press will engage the high shelf boost, indicated by LEDs. The second button turns on a -20dB pad and a second press initiates the safety channel — a builtin measure of clipping insurance which records a second channel of audio at -20dB down to catch unexpected spikes cleanly. The LED next to these is a dB peak warning light. Rode states a 30+ hour lifespan from the rechargeable lithium-ion battery. Supplied with the mic is the Rycote shockmount, a foam windshield and all necessary cables. The mic itself is built well and firmly slots into the shockmount. Weighing in at 94g, you’ll hardly notice the difference using it with a handheld camera rig or camcorder. ON THE PHONE
Phone usability is invaluable. Plugging it into my Android smartphone with a USB-C to USB-C cable yielded instant level in my recording app
PROS Works with almost any device Lightweight and rugged construction.
CONS Mic sound quality not incredible
and much better sounding audio than the built-in mics. In our editorial line of work we rely heavily on our phones to hastily record interviews and conversations with people. Video is also a big part of what we do, for which we use ‘proper’ cameras. I love that I can plug the VideoMic NTG into my phone to capture an interview, then move it straight to a hot-shoe to record audio on a professional camera. Just as easily I can take it home and plug it into my iPad to put down a quick recording of that song I’ve been writing. In today’s multi-device world, versatility equals bang for buck — and the Rode VideoMic NTG covers all bases. As for the mic’s sound itself, it probably won’t blow your socks off. Especially if you’re a location recordist relying on purity and pristine signal paths. The mic's self-noise (15dBA) is comparable to other shotguns of similar breed and it responds well when paired with a quality, quiet preamp. Tonally the VideoMic NTG has a slight low-mid lump that I felt compelled to scoop out more often than not — but that’s me nitpicking. The tight supercardioid polar pattern does an excellent job rejecting sound to the rear and sides of the mic. Overall, for capturing clean audio off the top of a camera, you can’t go wrong.
SUMMARY Rode continues to impress with its innovative product design and problem solving. While you may find a more pristine sounding shotgun mic out there, you’ll be hard pressed to find one more versatile. VideoMic NTG is a fantastic choice.
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Producers’ Show 'n' Tell with:
Dave + Sam 1. Macbook Pro
2. Shure SM7B
Without this there’s no record! Everything was recorded, arranged, mixed and processed through this thing. Purchased in 2014 and still going strong!
What better selling point is there than it being the mic that “was used to record Thriller”. We recorded about 80% of No Shade on it...
Artist: Dave + Sam Album: No Shade
5. Yamaha Reface DX This thing is small but mighty and has some really special patches on it. Absolutely my favourite punchy bass sound, plus a lot of great smooth sounds for leads and chords. The original Yamaha DX keyboards were used extensively in early house music productions, so this thing is really great as a time capsule to capture that vibe and emotion.
3. Arturia DrumBrute This was more of an inspirational piece than anything. It made it on the record in a few instances, but more often it was used to get the energy going when we first started a session. Sam would start creating a beat and Dave would begin freestyling and it got us in the mood to create. It’s so fun and intuitive to program and the built-in FM synth module is really fun for jamming.
4. Soundtoys EchoBoy Plug-in This is our favourite delay plug-in — in general all of the Soundtoys plug-ins are amazingly creative tools but this one takes the cake. It was used for almost all the delays on the album, but it’s a real secret weapon as a tone tool, too!
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FEATURE FEATURE
With I,I, Bon Iver’s Justin Vernon finds yet another way to tell a story. Co-producer Chris Messina and engineer Zach Hanson fill in the blanks. Story: Paul Tingen
Artist: Bon Iver Album: I,I
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Justin Vernon, leader of the alt-folk band Bon Iver, called their 2016 album 22, A Million, “a new way of telling a story.” It was an understatement. 22, A Million was as experimental as it’s possible to get (short of going atonal), with tons of distortion, drop-outs, and weird sonic artefacts. The combination of disorientating sonic weirdness, gorgeous melodies and crazy song titles proved enormously successful, both critically and commercially. 22, A Million was a major achievement, but, as is often the case with artistic high points, where could they go from there? Getting even weirder risked severing their connection with popular culture, while a more straight-up album might be considered a retreat. The follow-up album, I,I, released in August, provides the answer. In many ways it is a return to more traditional songwriting, with longer tracks and more traditional song structures. It’s still awash with sonic experimentation but in this case the production weirdness is never so dramatic that it pushes the songs to break down, and Vernon’s voice sounds more natural and is more centre stage. With the novelty of the extreme production experimentation gone, reviews of I,I were slightly more mixed, while remaining overwhelmingly positive. One reviewer had it right when he called I,I “a moment of consolidation,” and concluded that the result is “a mature masterpiece and a stunning marriage of ambition and technique.” Another interesting difference between I,I and its predecessor is that many of the performance and production credits on 22, A Million were deliberately obscure, with names like ‘Noble Black Eagle’, ‘Trust Ear’ and ‘Scream Defence’. The credits for I,I, in contrast, simply list Chris Messina, Brad Cook and Justin Vernon as coproducers, while everyone else gets instrument, writing and/or technical credits. While Messina was credited as the ‘Maker’s Maker’ on 22, A Million, on I,I he’s credited as playing keyboards on some tracks as well as being involved in production, engineering and mixing. Zach Hanson, who had the least cryptic credit on 22, A Million with ‘Professional Mixer,’ has straight engineering and mixing credits on I,I. Via a Skype conversation from Eau Claire in Wisconsin, co-producer Messina and engineer Hanson gave some insights into the making of I,I, inevitably harking back to 22, A Million as the blueprint for the band’s current direction. MAKING I,I
“There was no over-arching theme for the direction of the new album,” recalls Messina. “There was a larger emphasis on collaboration with various musicians and producers, but we did not set out to do something radically different. We weren’t saying, ‘the last record was too this or too that, let’s do something else.’ In fact, a couple of the songs on the new album, ‘iMi’ and ‘Faith,’ were from the 22, A Million sessions, and we simply continued from there. We simply went back into the studio after 22, A Million was released, and started messing around.” “Justin’s voice is something we all agreed
was going to be less fucked up this time. We realised there was an opportunity for Justin to say something much more clearly than he has in the past, and to speak less abstractly. That meant we were putting a lot of emphasis on lyrics and the sound of his voice, and it’s louder in these mixes than it has been previously. Those were conscious decisions. There are still places where his voice is messed up, but the effects are not as front and centre as before.” About six years ago the first steps towards 22, A Million came out of Vernon hanging around depressed on a Greek island, manipulating a vocal sample on a Teenage Engineering OP-1 so it ended up sounding like ‘two-two’, and realising that his new direction was going to be about numbers and experimentation. This resulted in a lot of musical soul-searching that continued during three years of sessions at April Base, his studio in Eau Claire. With I,I things were very different, recalls Messina. “We were aware earlier on that we were actually making a record, and Justin was in a very different place mentally. There was more focus on confidence with the new album. That showed throughout the entire process of making I,I. We were very focused, and everything just made sense for the most part. We were never about being weird for the sake of it, so we didn’t feel that we had to repeat that. It was always about the emotional impact. The next record could be even weirder than 22, A Million, or it could be Justin and a guitar. There are no rules, we simply go for what comes together and works.” APRIL BASE
Like for 22, A Million, the writing sessions for I,I took place at April Base. The importance of this location was illustrated by its ‘Producer’ credit on 22, A Million, which is almost certainly the first time a studio has received such a credit. Messina is the studio’s manager, and Vernon’s main sidekick. He first met Vernon when he worked for a live sound company called Eight Day Sound, based in Cleveland, and was a sound engineer for a Bon Iver tour. The two became friends. Vernon invited Messina to visit his studio; Messina moved from New York to Wisconsin in 2011. “At the time, the studio was just a Pro Tools rig and some Teletronix LA2A compressors, Neve 1073 mic preamps, and a few microphones. Justin said to me, ‘do whatever you want to turn this place into a proper studio.’ We started with Studio A, which was the original space; it’s been through five different iterations now. Over the years we’ve had tons of different gear coming in and out, including a Trident 80C desk and a Studer A827. We eventually decided to build a second studio, Studio B, which has an SSL Duality desk. We’ve also built several live rooms. The two studios and their live rooms ended up being totally tied together. Most of the writing sessions for I,I took place in the live room of Studio A and the control room of Studio B. Currently, the entire studio has been emptied for renovation; in anticipation of the renovations, we sold the Trident during the making of I,I and emptied the control room of Studio A. All
the outboard from that studio started piling up in Studio B; it was totally nuts!” According to Messina, the core team that worked at April Base was Justin Vernon, Brad Cook, Messina, producer BJ Burton, and Zach Hanson. Members of the live band regularly came in, as well as violinist and arranger Rob Moose – who also did the arrangements for the Worm Crew, a five-piece horn band plus harmonica player. More than 30 other musicians are listed in the credits for I,I including luminaries like James Blake and Bruce Hornsby. The various musicians, engineers and producers present at April Base played their instruments and messed with the enormous amount of gear there. One piece of kit that stands out in particular is the ‘Messina’, obviously named after Messina himself. It was the most prominent vocal effect on 22, A Million¸ and some reviewers misinterpreted it as simply AutoTune. According to Messina, his namesake is used more sparingly on I,I but is still an important feature.
We were never about being weird for the sake of it... It was always about the emotional impact.
MESSIN’ WITH MESSINA
The Messina was inspired by software designed by Francis Starlite (of Francis & The Lights) called the ‘Prismizer’, that can make one instrument sound like many. “Francis stayed at Justin’s house for a year,” recalls Messina. “Inspired by what he did, Justin and I tried every vocoder we could find but it always sounded like a vocoder. Instead, we wanted to keep the character of whatever input signal we used. We ended up sending a signal to Ableton Live and treating it with two instances of AutoTune. The first one tunes the vocal, the second creates a single note that is the tonic of whatever phrase is sung or played. That tonic note is sent to an Eventide H8000, which we set to a Midi-controlled harmony program. Justin plays a Midi keyboard and the H8000 receives what he plays along with the tonic note from AutoTune, and generates up to four notes that are, to some degree, random. The final Messina sound is a combination of the dry signal, the tuned signal, the tonic note, and the harmonies created by the H8000.” “On 22, A Million you can hear Justin on his AT 17
own with the Messina on the track ‘715.’ On the new album the chordal information in the tracks ‘Holyfields’ and ‘Jelmore’ is from the Messina. On ‘Holyfields’ it was a bit of a process: we were messing with an EML Electrocomp 101 (an early ‘70s patchable synth) for five or so minutes, and then Brad started playing different octaves of D on a Yamaha Reface CS synth and looping them. We sent the loop to the Messina, and set up gates that were triggered by the EML so that everything was pulsing along with the EML. ‘Holyfields’ ended up being almost entirely one take; Justin sang on it immediately afterwards. For ‘Jelmore’ we had a similar process, but with a weird loop of one note created on a Jupiter 8 as its basis. It was really distorted; we looped it with long crossfades, so there’s a lot of movement, and then sent it to the Messina which was being played by Buddy Ross. We sent that to a Lexicon 480, and gated the return of the 480 with a prepared trumpet and also an Folktek Modified Omnichord to make it react in a really bizarre way.” It’s obvious from the above that messing with hardware is what turns Vernon and his fellow sonic travellers on. This extended to using a desk and outboard, as Zach Hanson explains, “Justin finds more inspiration in instruments and outboard that make a crazy sound, rather than throwing on a plugin. I would say that probably the most important gear addition for this album was that we got two Eric Valentine Undertone UnFairchild compressors. They are insane. We used them on Justin’s vocals, and we also had one on the mix bus.” AT 18
SONIC RANCH
The sonic journeys undertaken by Vernon and Co. also included a physical journey, from April Base to Sonic Ranch in Texas. The latter bills itself as ‘the world’s largest residential studio.’ Messina: “Apart from the UnFairchild, probably the biggest other change with this album was that we went to another studio to finish it. The huge renovation we had planned for April Base started in January of this year, so we had to move elsewhere. We were in Sonic Ranch for six weeks during February and March, and we took over almost the whole place! We were using five of their six studios for the whole time we were there. Zach and I were mixing in one studio, which was called ‘Adobe’. Justin set up in a small studio by himself to do vocals and work on lyrics. The band was in two tracking studios. We were also scoring a film and had set up a studio just for that. We brought in a truck full of our own gear – mostly the weirder stuff that other studios don’t have, because Sonic Ranch has more of the staples than you’d ever need!” “By the time we arrived at Sonic Ranch the songs were in a recognisably finished state, apart from one or two tunes. Justin finished all the lyrics and his vocals at Sonic Ranch. Musicians and producers came in for a few days to a couple of weeks, adding finishing touches or sometimes coming up with new things. There was a lot of hanging out!” “One recording that we’d been struggling with for six years consisted of Justin and his best friend Trever Hagen messing with a radio in a barn. That radio had been there for a long time, and they
just turned up whatever was on it. While they were doing that, they were also sliding pieces of cardboard across the dirty concrete floor in the barn, and recording everything on an Android phone of some kind. That sample had been at the core of a track we called ‘Barn’ for years, and we’d been trying all of that time to turn it into a song. It was in that place where you love it but then you hate it, you’re excited by it but then you’re frustrated by it. Andrew Sarlo, a producer, came to Sonic Ranch to work with us, heard ‘Barn’ and said he wanted to mess with it. We heard enough in what he had done to offer us a new way to finish the song. It’s like he had created the first chip in this huge wall that we’d been staring at for years; after that we quickly blasted through it. The sample is the opening track of the album, ‘Yi,’ and it runs into the tracks ‘iMi’ and ‘We’.” RECORDING VOCALS & MORE...
Zach Hanson has worked regularly at April Base as an engineer and mixer since 2012, when he met Vernon while working as the drum tech for Sean Carey (Bon Iver’s drummer). At April Base he mixed The Staves’ EP If I Was (2014) and album Sleeping In A Car (2015). Both releases were produced by Vernon, who was so satisfied that he offered Hanson the job of mixing 22, A Million. Hanson was involved from the beginning with I,I, recording several of the sessions and then mixing the entire album with Chris Messina. Hanson and Messina discussed some of the recording signal chains, beginning with Vernon’s
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JELMORE To illustrate their mix approach for I,I, Hanson and Messina chose ‘Jelmore’ as the most indicative. The first six tracks are sends to outboard gear – an Eventide H3000 side chain and a Thermionic Culture Vulture side chain. Below that are nine drum tracks, four keyboard tracks, two sax tracks, a bass track, four Messina and H3000 tracks (returns of the top six tracks), then vocals. At the bottom are a number of group/stem tracks that are prints of signals coming back from the desk. Messina: “The six tracks at the top are duplicates of other tracks in the session. These were edited to taste and then used as the key input sources for various gates and sidechaining. They each triggered the key in a different way and at different times. Hanson: “We had a Folktek Modified Omnichord that created some wild sounds. The EBass track that’s sent to the Culture Vulture comes up in the bridge for extra colour. The drums below it have very little treatment because they just provide motion. Below the drums is ‘buddy juno’, which is basically bass sub swells. We added the Waves LoAir Subharmonic Generator to get crazy low energy. Next is the modified Omnichord, with a FabFilter Pro-Q2 as a low-pass filter. Below that are a piano track and two sax tracks. Chris added the SoundToys Little MicroShift to the bass track to create a warble and chorus effect and make it more stereo. For some reason, that particular part was really cool spread across the stereo field.” Messina: “The bottom two of the four pink tracks – the H3000 and the Messina – are the core of the song. The H3000 sound has the Waves S1 Stereo Imager automated to open up in the bridge. There’s a Messina Octave track pitched an octave down from the original Messina track with the Waves SoundShifter to give the bridge some more weight. The Messina Culture Vulture track below it fades into the second verse for some grit, then disappears again. The UAD MXR Flanger/Doubler provides separation from the dry Messina track.” There are six tracks of ‘ooohs,’ sung by Sean Carey; they only have the Avid BF76 compressor. The four ‘Waz Vox’ tracks were not used. Next are nine Justin Vernon vocal tracks, with the lead vocals having the FabFilter DeEsser, Antares AutoTune Pro, and the FabFilter Pro-Q2. Messina: “This is a rare song in which we used AutoTune throughout the song. It was set very lightly, just adding the AutoTune sound.” Hanson: “What you do not see in this session are the sends and returns from the outboard like the AMS reverb and the Hawk delay. We used the AMS on all voices. Below Justin’s vocals is a ‘Jelmore’ mix print track, and then three scratch vocal tracks that we did not use, and finally the stems, which we printed post-fader from the desk, for recall.” Hanson: “The Neve 8088 desk has four master buses, and we used all of them. One and two were the main mix buses with the Undertone UnFairchild compressor and Maag EQ4M mastering EQ, the latter generally with +0.5 to +1dB boost at 2.5kHz and we played with the Air Band at 20kHz and 40kHz. We sent everything but the drums to the other two buses, compressed with a pair of Urei 1176 Blackface compressors, and mixed that in to taste; basically it was parallel compression. We never used any limiters; our final mixes were very much on the quiet side. Trying to be as loud as possible was not what this project was about!” AT 20
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voice. Messina: “We generally default to the Sony C37 because he just sounds incredible on that mic.” Hanson: “In some cases we used the Shure SM7 because Justin is super familiar with it. Sonic Ranch had a Neumann U47 that we’d set up to record guitars, but when Vernon sang into it we were like, ‘wait a minute, that sounds great!’ There were a few songs on which we tried both the C37 and the U47, and his voice sounded insane on both of them. So we ended up using those two mics the most. The signal chain was the mic into a Neve 1073 preamp, and then the UnFairchild.” Messina: “We also often used an AMS RMX16 reverb, plus the Hawk HE-2250 stereo tape delay. It’s a Japanese consumer product from the ‘70s, and it sounds ridiculously cool. It has a really pure sound; we fell in love with it and used it on many things. It’s used on vocals a lot, but it’s all over the album.” “We recorded Justin’s acoustic guitar with a U47 mostly through a Neve 1073 but sometimes direct into the Neve 8088 desk at Sonic Ranch. Keyboards generally were straight in. At April Base we have many Phoenix DI boxes. They make an eightchannel rack-mounted DI that sounds fantastic. From there we often went into a 1073.” Hanson: “I recorded the Worm Crew at April Base, live, with everybody in the room. There was a saxophone, clarinet, bass clarinet, two trombones, a trumpet or French horn, and the harmonica. I had spot mics on each individual AT 22
Signals such as a piano... had so much more energy and warmth when sent through the desk
performer and then a stereo room set-up using two Flea 49 mics, maybe 20 feet away from them, pretty high up. I also had two Flea C12 mics as a close stereo pair, and a ribbon AEA R88 as well. ‘Marion’ was a live take with Justin playing guitar and singing with the Worm Crew playing, all of them in the same room. We usually don’t want spill, but we certainly didn’t mind it in that scenario – in fact, spill was the purpose!” MIXING
As is common in these days of working with DAWs, Messina and Hanson mixed the sessions as they were recording. But, asserts Messina, “We had a final mix stage in Texas. The way things were structured is that Zach, Brad, Justin and I arrived 10
days before everyone else to work on the sessions. Then all the musicians and producers showed up for a couple of weeks, and then we had another 10 days during which we finished everything. Zach and I mixed, but the others had a lot of input. Once again, there was a lot of hanging out!” [laughs] “The final mixes usually involved Zach doing cleanup, getting rid of all the pops and clicks and getting all the tracks in order. We mixed through the 40-channel Neve 8088 desk at Sonic Ranch, so while the sessions stayed more or less the same in Pro Tools, we created stems and assigned outputs. All the detailed nit-picking changes happened in-the-box, as did the automation – mostly using clip gain. Although the Neve desk has Flying Faders, we only used it to take snapshots of each song, not for automation.” There’s still a lot of romance attached to mixing on a desk, but as anyone honest enough will admit, it’s a challenge. It’s much easier to load up a session in Pro Tools, so why mix on a desk in 2019? “It was mostly for the sound of the desk,” replies Hanson. “That became apparent almost immediately. Signals such as a piano recording that we’d been hearing for a couple of years suddenly had so much more energy and warmth when sent through the desk. It was mind-blowing for Chris and I. We were like: ‘Holy smokes! This actually does make things sound better!’ Hearing things come back sounding more alive and interesting was really exciting.”
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FEATURE
Michael Bublé Live He’s one of the biggest touring acts in the world and the audio production one of the most highly finessed. Story: Christopher Holder Photos: Preshan John
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‘A mix is a mix is a mix.’ This is the sage advice given to FOH engineer Craig Doubet by a recording engineer mentor David Leonard (who won a Grammy in 1983 for co-engineering the Toto IV album). By which he meant: a good mix will translate across any monitoring system. Craig carried that lesson into his live sound career. In a live context, the lesson is: make sure your well-balanced mix sounds good in the room by getting your PA right — right design; right tuning. Which is why Craig Doubet personally oversees the intricate Michael Bublé PA design. He knows most of the arenas in the world pretty intimately and armed with Meyer Sound’s MAPP acoustic prediction program, will make sure his crew chief knows exactly what the PA design will be as they back the trucks up. And what a PA it is. MEYER: L OF A PA
The Solotech-sourced PA is entirely Meyer Sound and mostly ‘L Series’ with some Mica. A 200+ box Meyer PA is always awesome to behold but the special sauce is in the use of a Meyer Spacemap beta that allows Craig to move Michael Bublé’s vocal image down the room when he ventures onto the catwalk to the B stage. The design introduces additional arrays that carry only the vocal and help localise Michael’s vocal when he’s on the B stage. I’ll allow Craig to explain further: “Michael started using a B stage about 10 years ago. He loves it — he loves that connection with the audience. On the last tour, we had a large B stage surrounding the front of house mix position and we even had punters inside the stage — it was a real feature of the show. But depending on the venue, the B stage/FOH position could be as much as 60 metres from the front of stage. As you can imagine, that’s introducing quite a delay between him and the output from the FOH PA. “Michael uses IEMs but he also uses a set with ports to get an ambient feed into his ears. What’s more, in that previous tour’s show the B stage at times featured a full band with seven open microphones. All up, this was introducing quite a slap off the PA compared to his ears. If you’ve never tried to sing or talk with a 100ms-plus delay, let me tell you: it’s not easy. I don’t know how he did it. “He always asked what we could do to fix it and really the only answer was: hang another PA. Which was never going to happen in the middle of that tour. “For this tour, we had a shot at addressing the problem. “This tour’s stage design sees a large bandshell and a long 30m thrust ending in a 10m-diameter B stage. Just like last time, he loves it and is always on the runway. “So my design goal was: ‘How do I cover the room without aiming any speakers at that thrust and that B stage?’ First up, I place two main hangs in the room where the B stage ends, which covers the back of the room. The advantage of that is, now I’m not throwing 100m with a main PA, I’m throwing a maximum 65m into the back seats. So
I reckon every live sound engineer should sign up to a Hippocratic oath: ‘d o no harm’
right away I’ve just decreased the reverberation distance in the direct field. “Next, we took a fresh look at what I’d have previously called our main hangs. This time I’ve got them pointing away from the centre of the room by about 20 degrees to keep their dispersion away from the thrust. Next to those is another hang that points straight out sideways. So between those three hangs aside, we cover the entire room.” B STAGE PA
Where it gets a little trickier is when Michael comes right out to the B stage. To avoid the issue of the last tour where Michael was hearing himself from the main PA 100ms or so after he’s singing into the mic (and hearing himself in his IEM), Craig takes the vocal out of the main hangs. More than that, to ensure the audience isn’t losing any of the vocal and hearing the vocal localised at the B stage (rather than a mis-match of hearing the vocal emanating from the main stage), two additional arrays are brought into play — one that fires back at the audience from the B stage and another that fires sideways. “So essentially his voice is moving where he physically is,” explains Craig. “Everyone still gets his vocals. It just changes perspective. If you’re sitting to one side, you feel the vocal shift because it’s removed from the FOH PA and goes to the side hang in the middle of the room. And that’s how we are using Meyer’s Spacemap on the Galaxy matrix. “Meyer built me an interface and a program that’s a virtual fader on an iPad. It has one job to do: to move Michael’s vocal back and forth. It was an enormous amount of work to set up but is now easy and reliable to use.” SPACEMAP BETA
Most of the time Craig is ‘panning’ only Michael Bublé’s vocal up and down the room. Does the vocal slide a little with the timing of the rest of the mix? Yes. But as Craig points out, that’s not such a big deal for this genre (just listen to some old Frank Sinatra records to hear how elastic the timing of the main vocal can be). At one point in the show, though, Bublé plays an ‘intimate’ club set at the B stage with a small band. At this point the whole B stage band is being reproduced through the B stage PA, with nothing coming out the FOH PA.
MONITORS: SIDEFILL, WEDGES & IEM The current tour saw Michael Bublé move from Sennheiser’s 5000 Series to the 6000 Series. The system is interesting for the fact it converts the signal to digital in the transmitter. I spoke to Monitor Engineer Marc Depratto about his experiences: “It’s been very stable. It’s given us more flexibility, in the sense that we can stick the transmitters very close to each other and you don’t run the same risk of intermodulation. Which is especially true of a show like ours where you’re in 20-plus countries in year. We could be in Dallas one week — which is an RF hell-storm — and then in another city with a highly regulated RF landscape and not have any problems. It makes it much easier. “The 6000 series also sounds great. That said, I do miss the KK104 capsule. The KK204 capsule is great but I miss the low-mid response of the 104, which was really apparent on a big powerful voice like Michael’s.” Michael’s monitoring setup is interesting. He has ported IEMs as his main source but that’s supplemented by sidefills and wedges. Marc explains what’s going on: “Michael likes to feel his voice — those rich, thick lows. The wedges are running particularly loud but they help supplement the in-ears, the spill from the PA, and the band itself. The combination serves to envelop him. “Other than that, I aim to give Michael a full mix. He like the mix to breathe — he’s not interested in too much compression and will call it out as soon as he hears it. And the mix needs to sound sympathetic to what he’s hearing behind him from the band itself — he’s got to be able to pull his in-ears out and for what he hears not be jarring. “Would I recommend this kind of approach to monitoring to any other artist? I wouldn’t if you’re on a super-loud rock ’n’ roll stage. It wouldn’t work, you need isolation at the expense of everything else. For this kind of gig, it works, but it takes quite a lot of care and attention.” AT 25
SPACEMAP PANNING
SIDE ARRAY 11 x Lyon • Main Stage Mode: Off • B Stage Mode: Vox only
REVERSE B STAGE ARRAY 8 x Leopard • Main Stage Mode: Off • B Stage Mode: Vox only
B STAGE MAIN PA/DELAY ARRAYS 5 x Leo over 12 x Mica 6 x 700HPs a side. • Main Stage Mode: Band & Vox as Delay Array • B Stage Mode: Band & Vox with Vox Time Corrected for B Stage
MAIN ARRAYS 8 x Leo over 8 x Lyon 6 x 1100 LFC. • Main Stage Mode: Band & Vox • B Stage Mode: Band only
FRONT SIDE HANGS 14 x Lyon • Main Stage Mode: Band & Vox • B Stage Mode: Band & Vox with Vox Time Corrected for B Stage
FOH Engineer Craig Doubet mixes the Melbourne show on his SSL L500 Live.
Of course, Meyer has been doing this kind of sound design gymnastics for years, working with musical theatre and Cirque du Soleil on some amazing immersive audio productions — mainly using its DMitri platform. But this was taking the Spacemap software and applying it to the Meyer Galaxy matrix. “It just made life easier,” recalls Craig. “We didn’t need to get another piece of gear involved into our control rack.” AT 26
SSL MIXING
The latest tour saw an appreciable increase in the number of inputs from stage, and Craig Doubet was looking to upgrade his mixing console as a result. He decided on a SSL L500 Live. He explains why: “I’m an SSL user from way back when — 1985, in fact, with the E series and G series studio consoles. When SSL came out with a live console they contacted me to see if I was interested in
taking a look. I liked what I saw and took a console out with Selena Gomez a few years back — learnt it and loved it. “The first thing I noticed is it sounds good — it sounds like an SSL to me, and that sound is in my DNA. Because I’m so familiar with the sound, it performs entirely predictably for me — when I tweak that pot it does what I imagine it’ll do to the sound.
B STAGE MAPP PLOT
MAIN STAGE
Meyer built me an interface on an iPad. It has one job to do: to move Michael’s vocal back and forth
B
MAIN STAGE
B
MAIN STAGE
MAIN STAGE MAPP PLOT
67'-1"
B STAGE
B STAGE MAIN PA/DELAY ARRAYS
REVERSE B STAGE ARRAY
MAIN ARRAYS FRONT SIDE HANGS
“The preamps are amazing. They are super clean, and have super high dynamic range. I run most of my inputs really hot and they sound amazing. “The layout of it appeals to me as well. There’s always at least three ways of achieving something and you get to choose your favourite way of doing anything. When I go back to other consoles now, I feel limited — ‘oh, there’s only one way of doing that?’. “The L500 console compression is great. The channel compressors are a model of SSL’s classic console compressors with all the controls you’d expect on a pro console. Then there are three onboard plug-ins available, which are amazingly beautiful. There’s the bus compressor SSL built for the E Series consoles that every mix engineer loves. SSL’s talkback compressor also comes from that E Series console, and remains the benchmark comp if you want to trash your source. “I also run a couple of compressors from a Universal Audio Live Rack that runs seamlessly via
SIDE ARRAY
the console’s MADI I/O. “I have moved away from using a lot of external effects. But, then again, UA produces beautiful renditions of its own compressors. So I’ll always use an 1176 on the upright bass, for example — nothing else sounds right to me. And I’m using a beautiful version of an LA-3A on Michael’s vocal — it’s super fast and works amazingly well live.” VOCAL CHAIN
Michael Bublé’s vocal signal chain starts with a Sennheiser 6000 Series microphone with a Neumann KK204 head. The 6000 system digitises the audio in the transmitter. The audio from the receiver goes straight to the SSL stagebox via AES and then splits to the two Digico SD7 monitor consoles — all consoles take the signal digitally. Once the vocal hits the SSL L500 Craig uses the console’s flexibility to massage it into shape. Craig Doubet: “One of the cool things about this SSL console is you can reorder the channel
structure. In the case of Michael’s vocal I put the high-pass filter at the top and then a dynamic EQ. From there I apply an external EQ compressor deesser, then the channel EQ and the channel limiter, which is there just in case. “The dynamic EQ does most of the legwork. I started using the BSS DPR901 [dynamic EQ] around 30 years ago and the SSL dynamic EQ serves the same purpose. It’s not so much of a de-esser in this case as a way of evening out the proximity effect when Michael moves his mic around — and he moves it a lot. He’s also a very loud singer; he’s got a big voice and can get away with moving his mic more than most people but I need the dynamic EQ to even out the frequency response. “The channel EQ is just notching out a few frequencies. Michael still uses floor monitors and some sidefill as well as IEM so there is some ‘ringing out’ of certain frequencies. “Overall, I try to keep the vocal as clean as possible. My starting point is to ensure Michael’s AT 27
MICS ON STAGE There are 38 musicians on stage, including a 13-piece brass section and a 16-strong strings section who are sourced locally. DPA microphones have been Craig’s favourites on strings and brass (4099s in the main). Drums use a combination of Shure, Neumann and Audix mics. All the DIs are Radial (“a great Canadian company”). In all, there are 98 inputs for FOH.
vocal sounds like Michael Bublé.” Craig then applies some light use of reverb to the vocal — a Lexicon 224 emulation and an EMT250 plate reverb emulator. “I love the EMT 250 because, to me, it’s always worked on a male voice. It’s not so much a reverb as a sense of space. It gives you some nice balance at the bottom register of the voice. “I have both effects in there all the time. Their proportion varies depending on the song.” MIX TECHNIQUE
The Michael Bublé show is essentially the same format as the Rat Pack shows of the ’50s and ’60s. Michael is the raconteur with the ‘amazing pipes’, charming young and old. It’s a show with potentially dozens of open mics, featuring some pin-drop detail counterpointed by huge orchestral hits. It’s a mix that requires painstaking preparation and finesse. Craig Doubet explains: “I reckon every live sound engineer should sign AT 28
up to a Hippocratic oath: ‘do no harm’. That’s my first responsibility, to do no harm to the music. I like to keep it clean — every input is as clean as I can make it. I get rid of any extraneous frequencies or bleed as best as I can. Or if bleed is unavoidable, I’ll use it. For example, I have 16 mics on strings, surrounded by reflective surfaces. As a result, I’ve learned that as I bring strings up, I can reduce the vocal FX send, because those mics are acting as an introduced ambience. “In songs where the strings or brass aren’t playing, they’re muted in a console snapshot. Some snapshots even have some level changes. Anything I can do to reduce bleed and have an unnecessarily open mic, I’ll do that with a fader move or a snapshot. “Every instrument has some compression. Sometimes to just save my skin in case of an unexpected level change, while at other times to help deliver that traditional big band sound. “Audiences have an expectation of what a big
band sounds like, stemming way back to those original vinyl recordings from the 1940s. Those pressings had quite a limited dynamic range, so you need to replicate that somewhat to make match expectations. “That said, thanks to this huge band, I’ll have 30dB jumps in level on occasion. It provides great impact and that’s what Michael wanted — he wanted the band to impact people; for there to be a real sense of power. “Above all, it’s about making sure the audience hears Michael Bublé. I constantly ride his level and ensure he’s sitting high in the mix. The beauty of this style of music, is when the band is really going for it, that’s not when Michael is singing, so I rarely have to choose between Michael and the band in the mix. “When I mix, I’m totally dialled into what I’m doing, and I get physically involved. I like to put some emotion into what I do.”
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TUTORIAL
In part three of this series about mixing Synth Pop, mixer/engineer Tristan Hoogland describes his processes for mixing synths.
In the previous two parts of this series we went through my processes of mixing drums and mixing bass for synth pop. Having established the foundation of our mix with the drums and bass, we can now begin introducing the more atmospheric and harmonic elements: the synths! Of course, the word ‘synth’ could refer to any number of keyboard sounds – it could be a warm pad from a Juno 60, a stabby chord from a Prophet, an aggressive MS20 lead line or even just an electro piano sound. Whatever the case, the various techniques described here can be applied in all sorts of scenarios. Let’s get stuck in to it! EVERYTHING IN ITS PLACE
I request wet and dry copies of all mix stems, but I almost always use the wet stems and treat the dry stems as an insurance policy. There’s no use trying to recreate what they’ve worked on and lived with for so long! I’ll request them to print any filter sweeps they’ve applied, but I may break out the filtered sections to new tracks so I can have finer control during those passages. I spend my initial prep time working out if a synth sound is truly stereo, or if it’s just a mono synth with some kind of inverted phase effect that makes it sound wide but might cancel out in AT 30
mono. It’s usually easy to spot this; it will sound disorientingly wide in stereo, but will thin out considerably when heard in mono. My solution here is to monitor in mono (or center the pans) and, if there’s a problem, I’ll invert the phase on one side to see if that rectifies the issue. If that fails to improve matters, I’ll ditch one side altogether, go mono and recreate a stereo effect with something like a microshift. Artists and producers tend to be allergic to synths being panned out, especially if the rough mix doesn’t illustrate it. I believe this happens because when synths are layered in a particular way the relationship creates a block of sound – a unity that comes undone when you pan them off or try to break them away from each other. Don’t let this discourage you from experimenting, but, unless what you’ve panned out feels better and there’s a part that can balance it panned to the opposite side, you’re exposing yourself to an onslaught of “narrow that back in!” feedback from the artist or producer. What I sometimes do is pan two lines that
balance each other out, or in this example play a ‘call and response’ role. To make it less shocking, I’ve also sent these to a reverb with the pans inverted to retain some sense of stereo. This can be an effective solution if you’re wanting to achieve a bit more width and space. NOISE
Sometimes I’ll receive sounds with a lot of noise. This can be a result of poor gain-staging, noise induced from effects, or even just the patch of the keyboard. I’ll usually leave the noise in and assume it’s part of the vibe. If it’s having a negative impact, or the client requests for noise reduction, I’ll attempt to rein it in. If I’m unable to tame it effectively with low and high pass filters, I’ll resort to iZotope RX’s Spectral De-Noise. Put it first in
the chain, then select a small area of the track where the noise is present but there is no musical audio (a millisecond or two will do). Click ‘Learn’ and hit play so Spectral De-Noise captures the selected noise as a reference. Before proceeding further, set the Threshold to +6.0dB and Reduction to 0.0dB – this will act as our starting point. Now, with the music passage playing right through, we can begin adjusting the Threshold and Reduction in real time to find the sweet spot for removal. Start by bringing up the Reduction slider until the noise is completely removed from the signal. Next, bring down the Threshold until the noise begins to reappear. Having found the area to tweak from, finely adjust each slider in tandem until the desired amount of reduction is achieved. The relationship between these two parameters is integral and takes a bit of massaging to get it right. Once I’m happy, I’ll usually process this at a higher quality in AudioSuite. MAKING SPACE
I rely on carving with EQ – and particularly using high and low pass filters – when attempting to make synths work together in a mix. Using the bass track as my reference point, I’ll turn up the synth parts one at a time and focus on anything that’s blurring the low end. I find this happens because artists will often record synths full-range despite being used to only fill a certain part of the spectrum. This means they can have a lot of unnecessary low end when heard in the context of the mix. I’ll start by using a high pass filter and set it quite high, usually between 400Hz to 1kHz depending on the material, and then drag it down until I get back the integrity of the sound and leave out everything below it. It’s not unusual to be parking this around 200Hz to 400Hz, but sometimes even as high as 1kHz! Naturally, context and source is the variable here. Once I’ve done this I might need to focus my attention further up the spectrum and
rein in some mids or upper mids to steer the sound back closer to the original tone. At this point it can become a bit like a game of whack-a-mole as you continue to alter the tone, but you get the picture. Conversely, there may be an excessive amount of high end. It can sometimes be difficult to hear this and/or the masking it might be creating, but a spectrum analyser like the one in FabFilter’s Pro-Q3 can display any unwanted top end. I’ll either resort to Avid’s Lo-Fi – reducing the sample rate down to somewhere between 16000 to 36000 cycles – or use a steep low pass filter to remove it. In the example shown here, from the song ‘Tonight’ by Golden Vessel, the pad track from the DX7 had no real musical content above 2.5kHz and yet it had an insane amount of noise – presumably from some kind of onboard chorus effect, which is often the culprit for this type of noise. Beyond this I’m usually quite conservative, preferring to make space by using a few subtractive moves with EQ; cutting rather than boosting. With this approach there are some areas where you need to focus your attention, depending on the sound. If it’s a thick pad or piano sound, you might need to look somewhere around 100Hz to 400Hz. If it’s a buzzy synth, look around 2kHz to 5kHz. For something furry and fuzzy it’s more likely to be around 400Hz to 1000Hz. Of course, these areas I’m removing could be in the sound itself, or in the elements surrounding it. If subtractive measures haven’t taken me all the way I’ll find some areas to boost, usually in the midrange between 500Hz to 1kHz and between 3kHz to 6kHz – anything that will highlight a part or create some contrast against the surrounding elements. HARMONICS & SATURATION
I generally don’t reach for compressors unless a synth part is wildly dynamic; I find volume and corrective EQ gets the job done most of the time. If, however, I have a track full of synths that are
tonally similar I will experiment with distortion to achieve contrast. If a part is stabby or has a lot of transient, I’ll drive the signal with something like Lo-Fi, Culture Vulture, iZotope Neutron or similar. I prefer using distortion over compression in these situations because it rounds out the sound in a musical way and juices more out of the tone, as opposed to squashing it. Lo-Fi is great at quickly focusing a signal in a single move. I’ll grab the distortion knob, wind it up until it starts brick-walling during the loudest passages, and adjust from there (typically settling somewhere between three and six on the dial). Doing this might bring forward some unpleasant qualities that weren’t obvious before the processing, so I’ll have an EQ ready to address that. The Culture Vulture is great for fattening the tone and adding dimension to a sound; it drives and interacts with the signal in a way that’s akin to a guitar amp. It feels very musical with a broad range of tonal options. The ‘T’ setting is the warmest and most gentle of the bunch. I’ll usually start with this if I just want to take a sound up another 10%. The ‘P1’ setting on the other hand is great at adding aggression and fuzz to a sound; it’s good for brass pads and buzzy lead lines. With either the ‘T’ or ‘P1’ setting, I’ll usually set the bias between four and six on the dial and engage the 9kHz low pass filter because the resulting top end is generally unremarkable. On the track ‘Unconditional’ by Touch Sensitive I used it as shown here to bring more focus to the mid range. Decent alternatives include Radiator and Decapitator. On the more extreme end of harmonic generation we have guitar amps (simulators or otherwise). I’ll resort to these in the rare cases where I’m feeling a sound is unremarkable or in dire need of a makeover. In the example shown here I had a dull sounding Prophet playing a AT 31
repetitive three-note chord stab. In the mix I felt like the song had way too much of the same tones going on, so I used Softube Amp Room to add some harshness higher up and bring out the harmonics of the passage. There’s also SansAmp’s PSA1; it’s an oldie, but it works great too. DELAYS & REVERB
I’m a huge fan of expanding the dimension of a mix. Even in events where synths come in entirely wet, I’ll push them even further out into the ether. I like effects that aren’t noticeable or cheesy, and the best way to achieve that is to focus on the immersive qualities of the space rather than the tempo or sound of it. For pad sounds I’ll use carefully timed delays, and my favourite tool for this is Soundtoys’ Primal Tap. Despite being a digital delay, it’s quite dull and murky which is perfect for this application. I’ll start with something like a 1/16th or 1/8th division (maybe even dotted) and set the feedback to taste depending on the desired effect (for a long reverb-like effect I’ll go for 80% to 99%, while for a short room I’ll use 20% or less). To make it sound even more diffused I’ll unlink the two channels and offset each side, something like 10ms to 15ms in opposing directions. Place an EQ after it and roll off a ton of the high end until you can barely make out the delay, then back it up a bit (2kHz to 5kHz). I’ll also put a high pass filter (around 400Hz to 600Hz) on the return of the delay to keep the repeats from muddying up the low end. AT 32
The modulation knob is also worth experimenting with. Another favourite delay for achieving this effect is Soundtoys’ Echo Boy when using its Space Echo or DM2 settings. Conversely to delay, if I’m using reverb it’s because I want to hear it. My go-to for this is the UAD AKG BX20. Despite being a spring reverb it’s incredibly deep and dark sounding, which is perfect for placing synths further back into the mix. I’ll set the reverb time to taste and maybe include a short pre-delay if I want a little more separation between the wet and dry signal. I want to hear the reverb in the mix, but because the BX20 is quite dark in tone it needs a little help to translate. So I’ll usually send quite hot off the source, and I’ll run an EQ on the send into the reverb and another on the return from the reverb – both quite drastically! I’ll apply a high pass filter and a low pass filter over the send going into it (400Hz and 6kHz respectively) and then further EQ the return in a flavourful way with the Neve 88RS to remove a lot of low end along with some liberal boosting around 500Hz to 600Hz, and also something higher up between 3kHz and 6kHz to really highlight the tone of the synths. My other favourites for synth reverb include Lexicon 480L’s Hall setting, and Valhalla Vintageverb’s Concert Hall setting. + + Tip: When using delays and reverb it’s always worth experimenting with inverted panning to achieve even stranger stereo effects, particularly if the dry signal is panned to one side!
DON’T HESITATE, MODULATE!
Chorusing and modulation effects are great at keeping a signal audibly dry, but smear them into the mix so they feel less stark and upfront; this is perfect for synths that intentionally have no ambience effects applied. In my toolkit I’ll default to either the Brainworx ADA Flanger, Soundtoys MicroShift or the UAD CE-1. The ADA Flanger is incredibly versatile; it can be a way over-the-top warbling flanger, or a gentle and soft wavering chorus. I’ll usually insert this on the track itself, rather than via a send. I start by leaving the threshold full, turn the range quite high so I can exaggerate the effect, and then find the rate I want by adjusting the speed. I’m using this for tone just as much as I am using it for a chorusing effect, so I tend to go for something medium-slow and wavy (less than 3.00 on the alphanumeric display) that softens the sound and takes the edge away. Then I’ll dial back the range considerably and set the overall wet/dry mix to taste, generally arriving somewhere around 15%. UNTIL NEXT TIME...
That about wraps things up for this instalment. I also use sidechaining techniques when mixing synths, but I discussed those in detail in the previous instalment about mixing bass (see issue 136) and those same techniques apply to synths. Next issue we’ll delve into perhaps the most crucial part of mixing: vocals!
AT 33
REGULARS
PC Audio Plug-ins Under the Microscope Column: Martin Walker
Someone once said “a little knowledge is a dangerous thing”, and that is certainly true in the case of plug-in effects. With the advent of investigative utilities such as the excellent DDMF PluginDoctor (which I explored in AT133), and Christian Budde’s earlier VST Plugin Analyser, anyone can enter boffin mode and explore what their plug-ins are doing under the hood. However, it can be all too easy to write off a plug-in after misinterpreting test results. For example, one plug-in developer recently apologised on line for making plug-ins that completely confused most measuring plug-ins – as he jokingly pointed out, it can be like trying to run static test tones through a compressor and coming to a conclusion, when its effects on a dynamic signal like music are obviously far more relevant. So let’s take a closer look at how analysers can help, and when to take their results with a pinch of salt. ON THE LEVEL
One telling factor can be input level. Many plugins today offer analogue ‘mojo’ in one form or another, but, as many of us have experienced, the ‘sweet spots’ of analogue circuitry often occupy a small area between beautifully clean and nasty clipping. You may be able to ram a plug-in with a 0dBFS test signal and see good results displayed in an audio analyser, but, like their analogue counterparts, mojo plug-ins often expect a typical input level around -18dBFS to hit their sweet spot and will therefore be more successful when placed on individual tracks rather than across the higher levels of the mix buss. Many helpful plug-in developers specify an expected input level, or provide a level meter or overload indicator to help you send suitable levels into their plug-ins, but not all do. So, if you’re not sure if a particular plug-in is suitable for mix buss use, it can be helpful to run some analyser tests to check what your ears should already be telling you. For example, I often use various popular mojo EQs to add a certain something to the high end. Some, such as the famous Maag EQ2 and EQ4 plug-ins, have an overload indicator that
AT 34
illuminates just below 0dBFS, and can therefore offer clean ‘air’ for channel or master buss use. Even with a hefty 0dBFS input signal, PluginDoctor displays a minuscule 0.001% distortion from those plug-ins. In comparison, Soundtoys’ characterful SieQ at its default drive settings measured around 10% THD with a 0dBFS input signal, and sounded really overloaded on the mix buss. With an input level of -18dBFS, however, the THD dropped to 0.15% and it sounded glorious. (If you’re determined to try SieQ across your mix buss, pull its drive control to minimum where the THD drops once again to 0.15%) READING BETWEEN THE LINES
Context is an all-important factor in analyser investigations, because if you’re not careful you can come to some unwarranted conclusions. For example, I recently put the aliasing content of one particular plug-in under the microscope. Aliasing is a particular bugbear of digital audio, because if the plug-in generates distortion products above the Nyquist frequency (half the sampling frequency) they end up ‘folded’ back into the audio spectrum and will appear as a set of nonharmonically-related vertical lines on the analyser. In other words, unlike normal harmonics whose frequencies tie in strongly with the original audio (as it does with analogue THD), digital aliasing products tend to stick out as harsh and edgy. Hence they have become one of the big nasties of digital audio. Here’s a useful tip: a careful choice of the sine wave test frequency can make aliasing products a lot more visible on screen as a second set of vertical lines. If you want to be ruthless at exposing aliasing products, try 993Hz instead of the more typical 1kHz for 44.1k sampling rates, and 996Hz for 96k sampling rates. This will place the frequency of any sets of aliasing lines in-between the existing harmonics, rather than jammed up against them and therefore almost invisible. With this particular plug-in and a 0dBFS sine wave test frequency of 993Hz I could see what appeared to be lots of nasty aliasing poking out
above the background noise (see screenshot). Notwithstanding, it is generally accepted that anything below -120dB is inaudible (it equates to a minuscule distortion level of 0.0001%), and all the nasty aliasing I could see occurred at lower levels than that, which I confirmed as inaudible. When I use PluginDoctor’s HarmonicAnalysis page, I zoom in to display 0dB to -120dB, instead of its default range of 0dB to -200dB, because anything displayed below -120dB isn’t going to be audible. DEALING WITH EXTREMES
The effects of many plug-ins may also be extremely frequency and level sensitive, so don’t get too hung up on what appears with one test signal – this may change significantly at other frequencies and other levels (try sweeping the test frequency for example). Also, remember to check frequency extremes when putting your plug-ins under the microscope. At the low end, look out for significant content at 20Hz and below (even DC may appear with a few plug-ins). You’re unlikely to hear this through your loudspeakers/headphones, but it will nevertheless show up in an analyser, and it could damage loudspeakers on large systems if ignored. If a plug-in is adding any significant content at 20Hz or below, put a high pass filter after the plug-in to remove it. On the other hand, don’t bother too much testing how plug-ins react to high levels of high frequencies – bear in mind that above 8kHz real world levels are likely to be at least 20dB down on the rest of the spectrum. Overall, remember the popular audio maxim ‘if it sounds good, it is good’, and always use your ears rather than your eyes when choosing plug-ins. Yes, it may be tempting to follow the herd and buy or sell a plug-in because someone praises or rubbishes it online, but if your ears aren’t sure then a few one-off analyser tests on a particular plug-in may provide all the confirmation you need. Making music is a constant learning process, and it would be foolish to abandon a particular plug-in simply because we are abusing it in ways its developer never intended or subjecting it to tests that are highly unlikely to happen with music.
AT 35
REVIEW
MEYER SOUND ULTRA-X40
NEED TO KNOW
Active Compact Loudspeaker
PRICE $9,999 RRP CONTACT Audio Brands Australia: (02) 9659 7711 or www.audiobrands.com.au
AT 36
PROS Point source fidelity. Controlled dispersion. High power-to-weight ratio. Handy size.
CONS No volume control. Limited I/O.
SUMMARY Meyer’s new Ultra-X40 is the result of decades of on-going research and improvement in the pursuit of technical excellence, offering a tightlycontrolled dispersion and high quality sound in a box that is surprisingly smaller than it sounds.
Meyer’s new Ultra series combines everything they’ve learnt from the UPA and LEO series into one small box. Does it do its predecessors justice? Read on... Review: Mark Woods
Meyer Sound Laboratories’ popular UPA speakers are retiring after a long and distinguished career as leaders of the professional mid-sized point-source class. The full-range passive UPA-1 was released in 1980 and was the first live speaker to use a trapezoidal cabinet – a shape that allowed for curved arrays. Output quality was maintained by a dedicated electronic controller containing active crossovers, frequency and phase correction circuits, and feedback circuits to provide driver protection. In the mid-1990s Meyer Sound embraced powered loudspeakers by putting the processing and an amplifier inside the speaker cabinet. The active UPA-1P has been delivering Meyer’s trademark high-fidelity-at-high-volume sound to appreciative audiences ever since. Mention them around sound types and you’ll likely hear an anecdote along the lines of “I heard this big sound and it was all coming from these little speakers”. Meyer UPAs for sure... The UPA’s replacement would have to be good, and it’s awesome. The new Meyer Sound Ultra-X40 is still a compact, vented two-way speaker but it’s a completely new design with several fundamental differences. Over the past few years of developing their LEO family of large-scale line array systems, Meyer made significant advancements in driver configuration, horn technology, low distortion and phase correction, and wanted to incorporate these in a point-source box. The timing was good to replace the ageing UPAs, and the result is a speaker that is smaller, lighter, tighter, louder and cheaper than its predecessor. Lots to like there. UP CLOSE
The biggest difference is the driver configuration changing from the traditional horn-above-woofer two-way layout to a clever concentric-like design. The 12-inch woofer has been replaced by two eight-inch neodymium magnet drivers mounted on a concave waveguide and slanted inwards. Mounted between them is the central rotatable waveguide for the three-inch high-frequency compression driver. Applying some array trickery to the eight-inch drivers and coupling them with the back of the horn waveguide delivers a coaxial point source experience with exceptional directional control. Dispersion for the Ultra-X40 is a wide but focused 110 x 50 degrees (the Ultra-X42 variation is 70 x 50 degrees) and that pattern is maintained down to a low 400Hz. The two eight-inch drivers have a collective cone area similar to a single 12-inch driver but they’re narrower and the horn is in between them so the Ultra-X40 is not only smaller but is also about
9kg lighter than its direct predecessor, the UPA1P. That’s a significant difference and broadens its potential applications – although there is no facility for using it as a floor monitor. It can fit into a smaller case, and it’s small enough and light enough for one person of reasonable strength to put it onto a stand. Meyer noticed users were making their own mounting brackets for the UPAs, but they shouldn’t need to with the comprehensive range of custom-made accessories for the Ultra-X40. The speakers are fitted with 11 x M8 threaded points, and options include a hanging yoke, u-brackets, cluster plates for locking speakers together, plus the pole mount. For horizontal mounting the horn can easily be rotated via four screws under the grille; it doesn’t need to be removed from the cabinet. The cabinet is made from multi-ply birch and the overall look is professional but understated. The textured finish looks like the finish on other speakers, as does the perforated grille, but the angled speakers and big horn behind the grille look purposeful and modern. There are no side or top handles, resulting in a clean look. The chunky handles on the back are good for general handling and protecting the connecting leads, but they’re not much help if you’re lifting it onto a stand. CONNECTIONS
The rear of the cabinet is mainly heatsink with a powerCON20 in and out. Connections are professionally minimal – an XLR in and out is all you get, with LEDs indicating power, signal and limiting. This is fine for large-scale productions and installations but for smaller shows a volume knob can be handy; I had one show where the speakers were linked together and I wanted to run them at different levels due to their placement. The only time I missed a power switch was powering down at the end of a long night when it seemed a lot to go around to the back of the speaker to unplug the power lead. The connection panel is modular and may evolve over time. At the moment an optional five-pin XLR version is available that will connect audio plus the RMS remote monitoring system for remote muting or providing a readout of the loudspeaker parameters via Meyer’s ‘Compass’ software. Meyer are one of the manufacturers behind the emerging Milan network protocol that seeks to combine the AVB standard with defined device requirements so the machines can talk to each other. The results of that could lead to a networked version. Inside the cabinet, Meyer’s ‘Intelligent AC’ automatically selects the operating voltage and conditions the power by controlling voltage peaks and interference, while their proprietary audio chain does its thing with the phase and frequency response – resulting in the flat response, coherence and impulse response that gives these speakers their special sound. Meyer make all their drivers in-house to ensure complete quality control and maximum integration between components. Power to the drivers comes from a new power-efficient, air-
cooled, three-channel Class D amplifier capable of 1950kW peak. Cabinet size may be down but volume is up – the stated linear peak SPL of over 130dB [measured free-field at 1m using Meyer’s M Noise] from a speaker weighing less than 24kg is class-leading.
The point source delivery and controlled pattern cut a slice of clear sound through the potentially boomy, washy room
IN USE
The Ultra-X40s proved to be a delight to use and a crowd-pleaser. The first time I used them was at my usual testing ground, the Theatre Royal Castlemaine. We’re already fans of the brand; the house PA is a double Meyer MSL-3 system (plus four 18-inch subs per side) and works well in the 400 capacity room. In some ways the similarly shaped Ultra-X40s are a modern, considerably shrunken version of the 40 year-old MSL-3s. The specs are almost the same in terms of frequency response and volume (and also in terms of coverage if compared with the narrow-throw Ultra-X42 variation). As an indication of progress made over the last decades, the Ultra-X40 weighs 24kg while the MSL-3 weighs 110kg each... plus amps. I often set up extra speakers for a front in-fill at the Theatre; sometimes to fill the hole, sometimes for testing purposes. The X40s are an ideal size for my in-fill, but my first impression was how easily they could fill the whole room and compete with the main system. The big PA had more woof and rumble, but the X40’s mid-range was clear and coherent at the back of the room. They worked well together and the big system sounded a bit hollow without the X40s. The band that night was Finder Lover Seeker, an act that needed the three female vocals to be heard clearly above the full band sound. Their mixer, Sarah Madigan, had fun and at the end said she thought most of the vocals were coming from the X40s. The next week the Theatre Royal hosted Paul Kelly performing with Alice Keath as part of his recent ‘Love Is Strong As Death’ tour. I was doing sound for the Victorian leg of the tour, and had a similar experience to Sarah at the front of house. Using the X40s as in-fill changed the sound in the room; they didn’t seem to interfere destructively with the main system, but added real midrange clarity. They don’t mind going loud (their level needed to be watched to avoid interfering with the stage monitors), but the quality invites you to turn them up and it’s like opening the throttle on a powerful car – they lift up and go. Best in-fills I’ve used by a good margin. AT 37
ULTRA OPTIMISED Mark Woods talks with Pablo Espinosa, Meyer Sound’s Vice President & Chief Loudspeaker Designer, about the Ultra-X40. Mark Woods: How do the Ultra-X40s maintain their dispersion pattern below the horn’s crossover frequency? Is there some kind of array behaviour with the eight-inch speakers? Or is it the wave guide they’re mounted on? Pablo Espinosa: All of the above, and more. When using a single driver, the polar pattern of the driver is dominated by the size of the driver and the frequency being reproduced – this is just physics. One of the huge advantages of using two drivers in this configuration is that the combined polar pattern of the two drivers can be controlled and optimised to well below their crossover frequency. In addition, the polar pattern can also be optimised to interact with the horn’s polar pattern to create an incredibly smooth transition at crossover, on-axis and off-axis. MW: How did the R&D from the LEO range contribute to the Ultra-X40? PE: We started with this driver configuration and horn technology 10 years ago, and took it to the next level with the LEO family. As the family grew, every new member brought innovations. All these innovations contributed to the Ultra-X40. To create a product like the X40, it’s not just a matter of ensuring that each individual part (drivers, enclosure, horn, amplifier, etc.) has ‘best-in-class’ performance. They all have to be optimised so that each part interacts perfectly with all of the others to create a result that goes above and beyond putting a bunch of OEM components together. That understanding and optimisation process is the most important contribution from the LEO family.
A big reverberant old church in Maldon, as part of the Maldon Folk Festival, was too easy for the Ultra-X40s. I used them instead of a stack of black boxes and won’t be going back; the point source delivery and controlled pattern cut a slice of clear sound through the potentially boomy, washy room. The condenser mics were beautifully detailed on the fine instruments and stayed stable when turned up to working level. An audiophile’s treat, they have the ability to draw your ear back through the speaker to the source. Every touch of EQ, effects or panning is clear. This transparency is a Meyer Sound trademark and, along with their ability to faithfully reproduce all styles of music, is the reason for their reputation as exceptional speakers for theatrical applications. The annual Mountaingrass Festival was tougher. A new location for the main concerts had around AT 38
300 people sitting in an odd-shaped, half-dead, fluttery conference room. It’s a Bluegrass festival and opening up the condenser mic centre stage is always a good test – and one that this room largely failed. Unfortunate dimensions and fittings meant it was not going to give any real volume before it wanted to feed back. We had a good modular black box system that usually has done the job (previously in a big marquee) and a pair of Ultra-X40s. We set them both up and, as well as a noticeable difference in audio quality, the X40s were much smoother off-axis and more resistant to feedback. I tried pointing the two systems in different directions but that didn’t last long; it was sort of louder and covered a bit wider, but the modular black box system was interfering and detracting from the quality of the X40s – so I turned the modular speakers off and ran the X40s all weekend. The room needed a couple of ugly EQ cuts, but after that the X40s provided a fuller sound than you’d expect from a pair of speakers on stands in a big, bad space. The bass doesn’t sit loud in the mix for this music but it has to be there, and those upright basses go low; I was pleasantly surprised at the depth and clarity with no sense of strain. Bluegrass bands have a low average volume but extreme dynamics, and the mics are wide open. When they hit hard the meters jump and that’s another good test for a system; many get harsh in this situation, but the Ultra X40s revelled in the peaks. The audience at these acoustic events are very aware of sound quality, and I was getting the right reaction from them – along with some surprise that it was all coming from the little speakers. It’s that UPA thing all over again. Back in the real world, the final test was loud bands with the Ultra-X40s and separate subs. I was already impressed by what a pair of X40s could do on their own, but bands always need more. A Christmas party at Lot19, an artist colony and centre of Castlemaine’s alternate culture, was the test I wanted: an outdoors gig with two bands and DJs. The headline act was Vibrasonics, a 10-piece funky dance band using 24 channels to FOH and six foldback sends. I’ve used and reviewed a lot of
mid-sized powered speakers, waiting to find one to replace my comparatively lo-fi but tough and loud double four-way black box PA. The X40s combined with subs are the first to do it, the subs need to be good to keep up with them (Meyer recommend their LFC-750 subs). With a big bottom end under them they handled the whole band easily and also proved exceptionally good for music playback, similar to studio monitors in their detail and the way they draw the ear to subtleties in the original recordings. The DJs were happy and at the end of the night, packing up on stage with the tunes still cranking, it was noticeable how quiet and nice the X40s sounded from behind. Great vibe and the speakers were part of it; I pulled the pin at 2:30am under some protest. CONCLUSION
The Ultra-X40 is a genuine general-purpose speaker that will get used in theatres, themeparks and stadiums on its own or as part of bigger systems. You could put it anywhere thanks to its combination of low width and the rotatable horn; it’s ideal for under balconies or filling holes. The size and weight makes it more suitable for portable use than the heavier UPAs, and although it’s expensive the sound quality justifies the price. Along with the sound quality, the Ultra-X40’s practicality and expected long-term reliability may prove attractive for small operators as well as hire/ installation companies. The Ultra-X40 is the product of continual improvement in the pursuit of technical excellence, and will prove to be a worthy successor to the venerable UPA series. Meyer Sound have long been proponents of self-powered speakers, citing simplicity, increased fidelity and controlled integration between amp and drivers. They knew they had a hit on their hands when they took record numbers of pre-orders from trade shows on release. The Ultra-X40’s point source design offers superior sound quality, its tight dispersion gives predictable results, and its lighter and smaller enclosures increases practicality and placement options. State of the art.
AT 39
REVIEW
KORG VOLCA NUBASS
NEED TO KNOW
Vacuum Tube Synthesiser
PRICE $269 RRP
AT 40
CONTACT CMI: (03) 9315 2244, cmi.com.au or sales@cmi.com.au
PROS Tube oscillator and suboscillator give a warmth and grit to the sound.
CONS Controls should go to 11, or even 13.
SUMMARY A diminutive but mighty little tube-enhanced bass synth that harks back to Roland’s indomitable TB-303 in a more appealing format.
Continuing their pocketrocket Volca range, Korg’s Nubass harks back to the classic TB-303 but in a modern format with tube oscillators. Review: Christopher Steller
My first drum machine back in the ’80s was a Roland TR-606. It was what I could afford at the time and it didn’t stay in its original form for more than one day — it soon had separate outputs and tunable toms, and it was quite unique. At the same time, I was never compelled to visit its bass accompanist friend, the TB-303, because of its lack of synth facilities and because the sliding and squelching sequencer part of it didn’t really appeal. Keep in mind that this was the early ’80s, so that distinctive style we now know as ‘techno’ or ‘acid house’ didn’t exist (and the Devil Fish mods didn’t come until much later), plus I had already owned Roland’s 101 keyboard from their System 100, Roland’s SH-1, the Steiner-Parker Microcon and a Moog Prodigy, so my experience with synthesis was a lot more than the TB-303 had to offer. It was supposed to replace a bass player, and it did that job about as well as the TR-606 did of replacing drummers! I don’t feel like too big an idiot for not buying one at the time as it was considered a commercial failure by Roland and production was discontinued in 1984. Funny, that! SOMETHING NU
The first eye catching feature of the Nubass is the blue bubble at the top of the blue/black front panel. It’s part of Korg’s continues use and development of valve technology using a new type of vacuum tube, the Nutube; developed in conjunction with the Noritake Itron Corporation, utilising vacuum fluorescent display technology. It’s a valve, Jim, but not as we know it. Nubass has a VTO (Vacuum Tube Oscillator) combining a standard sawtooth or square waveform generated by one Nutube, with a suboscillator that can be driven to saturation through a second Nutube to give a warmth and grit to the sound without needing external processors. The ladder filter has a great sound with a spiky resonance that really squelches when cranked. The filter section has a simple envelope generator with control of attack, decay and envelope intensity, and, with a function shift, the LFO’s rate and intensity. Accent amount is also controlled here. The final tone enhancers in the Nubass voice are the drive and tone controls, similar to an overdrive pedal, which can compress the sound to add distortion and adjust the highs. Next to these controls are the Tempo/Swing knob and the volume knob. After the blue bubble, the second standout I noticed on the Nubass is that it has Midi so, in
addition to being used standalone with its onboard sequencing (and syncing with other Volcas, of course), it can be played with a full-sized keyboard, or added to a collection of other modules to be sequenced by your favourite hardware sequencer or DAW software. All of the buttons and keys are multifunction, utilising the ‘FUNC’ button to select the second function of each key to choose waveforms for the oscillator and LFO, activate the step sequence functions, or utilise other important parameters of LFO targets, motion sequences, etc. The 16-step sequencer can be used in step entry mode or played in real time using the 16 front panel buttons as a keyboard. You can save a total of 16 sequences and use the chain function to play multiple (up to 16) sequences tied together. The Nubass sequencer has transpose, accent and slide functions to give the characteristic note transitions of the techno and acid house trademark sounds, plus the ability to randomise those three functions to give continuous variation in a performance. Add in Volca’s standard motion sequencing of knob movements, which can be stored within each of the 16 onboard sequences, and the possibilities of the Nubass become even more apparent. Physically there are, of course, Volca’s usual sync in and out and the headphone jack, a 9V DC input for powering, and a battery bay on the rear of the unit — yes, even with the Nutubes, the Nubass runs on batteries! Do I have any criticisms? One major one: the Nutube controls should go up to eleven, or even thirteen. As it is, they provide a comfortable warmth and a bit of dirt. I would expect an uncomfortable heat and some speaker tearing with this included technology. Continuing Korg’s trend of diminutive but mighty synth engines in the Volca range, the Nubass harks back to the style of that silver enigma (the Roland TB-303) in a more modern and, to me, appealing format. Why? Sonic flexibility, appearance, innovation and price.
The ladder filter has a great sound with a spiky resonance that really squelches when cranked
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REVIEW
DYNAUDIO CORE 7 Active Studio Monitors After a three-year hiatus, Dynaudio’s newest range of studio monitors show a return to Core principles.
NEED TO KNOW
Review: Mark Davie
PRICE RRP: $3299.50 per speaker
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CONTACT Amber Technology: ambertech.com.au or 1800 251 367
PROS Plenty of power. Well-tailored low end. Just enough DSP. AES digital input onboard.
CONS Hard to fault.
SUMMARY With the Core series, Dynaudio has stripped back from the DSP-laden Air series to release a professional monitoring solution that gets it right in the hardware. The company’s new Jupiter testing facility is already paying dividends.
Dynaudio’s Core series of studio monitors has that feeling of getting back to basics. It was as if Dynaudio went a little too ‘mad scientist’ with the previous flagship studio speakers, the Air series, and Core is the natural course correction. The Air series was a bit of a marvel, no doubt. It promised to be a seamless intersection of hardware components and software DSP right from the beginning. Drivers would come off the manufacturing line with a perfectly-matched DSP correction to accompany them. But perhaps the more DSP you add and the more obvious you make it, the less it feels like you can trust the physical hardware. What’s to stop Dynaudio hooking up a hand-folded piece of paper to a DSP chip and calling it a professional listening solution? There was nothing wrong with how the Air series sounded. At the time of its debut it was a revolutionary idea. There wasn’t much in the way of operator-controllable monitoring DSP in studios, so Dynaudio figured it would pack that power into a speaker. If there was an uncomfortable feeling about them, it was more to do with the bulging LCD screen on the front panel. In the long run, it posed two problems that cut its life span short. Firstly, digital screens don’t age well. As unbelievable as it sounds, I’m sure iPhone Retina screens will be the equivalent of a dot matrix in years to come. Unfortunately, the Air series displays were dot matrix. Secondly, others soon figured out that it’s much easier and more comfortable to make those changes from the listening position. These days, if you want to make elaborate corrections to your monitor’s response, there are hardware controllers and software plugins to do it. It no longer makes sense to access it from the front baffle of your speaker while leaning over your desk. FILLING THE GAP
Dynaudio discontinued the Air series in 2016 to make way for the next generation of speakers. First came the LYD series, which took a bunch of Air’s DSP smarts and packaged them in BM serieslooking housings at more affordable prices. With a gap at the top of its range for the last few years, it felt like Dynaudio might be focusing less on the professional market. The Core series rectifies that with three models – the two-way Core 7, and the three-way Core 47 and Core 59. There’s also a quad-driver dual sidefiring sub that extends down to 15Hz. The series features completely new looks and cabinet designs. It’s not revolutionary, but I immediately liked it. It’s a spattered black coating, like the sprayed on Dura-Tex-type finish you’d find on modern PA boxes. The front baffle is flat, corners rounded, with dimples on all sides for mounting spacers – though I’m not sure why you need them on the top. Mounting them upside down so the tweeters are on the bottom, perhaps? The Core 7, which I had for testing, hasn’t lost its Dynaudio look. There’s still the soft-dome tweeter, and the woofer is instantly recognisable as a Dynaudio model with its proprietary Magnesium Silicate Polymer cone and dust cap with dashed
outline. At almost 15kg per speaker, it’s a solid box for the size. The Core 7 ships with a removable three-pronged cover for the tweeter, like a stumpy version of the plastic platform that holds up the middle of a soggy pizza box. LOUDER & LOUDER
The Core 7 has onboard Class D amplification that provides 500W for the woofer and 150W for the tweeter. It’s a lot of power for a speaker this size. As well as the ability to adjust the input sensitivity to match your gear’s output, there’s a maximum SPL level control with settings of 88dB, 96dB, 100dB and 112dB. In my relatively small mix space of about 5m by 4m I had them set at 100dB and they still went incredibly loud – louder than is comfortable, but without breaking up. Smaller speakers will usually hit a threshold before it gets really ear-blazingly loud. The 112dB setting was too much for me. It’s nice to know you can dial it into a range that suits your room, but still have the ‘oomph’ to handle a bigger space if you move. Given Dynaudio’s solid history for longevity, and the Core 7’s power handling, I would say these will last ages in any nearfield application. Dynaudio is also promoting these as an option for immersive live installations, providing M10 rigging points on all faces except the front and rear.
The imaging is spot-on, there’s a super wide sweet spot, and they’re really consistent off-axis.
BUILT TO BE ACTIVE
Among connoisseurs of higher end speakers there’s a perception that if you were choosing between active studio monitors and passive speakers paired with separate amplifiers, the latter combination would be the best way to go. There’s logic to those claims: a separate amplifier doesn’t interfere with the spatial design of the box, an onboard Class D amplifier couldn’t possibly outperform a Class AB Bryston, and so on. But these days it’s harder to blindly stand by those arguments, given that active monitors are a tightly designed system of box, speaker, DSP and amplifier. If Dynaudio was to make a passive version of its Core series, everything about the driver construction would be different. The Core series uses Class D amplifiers from Pascal, and a 64-bit DSP chip from Analog Devices. The drivers and enclosures are all designed and manufactured from the ground up by Dynaudio,
allowing control of the entire response of the system – from optimising the driver’s suspension, voice coil and magnet structure to suit the enclosure and amplification, right through to final DSP tweaks. Dynaudio also built a new testing facility, dubbed Jupiter, which allows the R&D teams to measure and improve their prototypes quicker and more often. FOR AIRHEADS
While a lot of digital processing is going on, it seems that – in contrast to the Air series – Dynaudio is downplaying Core’s onboard DSP. It’s still there, but you access it via rear panel switches rather than front panel LCDs, and the settings are typically wide-band rather than phase-mashing minute corrections. On the way in, you can opt for analogue or AES digital connections, both on XLR connectors. If you opt for digital, you simply flip a switch to decide which channel of the stereo data stream the speaker should play – Left or Right. There’s no sample rate conversion onboard: the Core series supports native resolutions at all the major stops between 32k and 192k. If you switch the sample rates of your playback material, the system will automatically detect the change, mute the speakers and come back to life half a second later at the new rate. There’s also a BNC Word Clock input to keep everything in sync. Unlike the AIR series with their front panel LCD screens, the only front panel indicator on the Core series is a single LED. In normal operation, it’s green. At amplifier clip it goes red, input clipping detection sends it to orange, and thermal protection is signified by a pulsing red followed by a 6dB reduction in level. It also pulses green when Word Clock and AES lose sync, before shutting off. There are three frequency ‘taste’ settings, called Sound Balance, to slightly tailor your sound: Dark, Natural and Bright. This tilts the entire spectrum by 1.5dB at either end, using a minimal phase filter to either brighten or darken the overall response without introducing any phase anomalies. It’s a great control to have. Rather than getting caught in a rabbit warren of separately adjusting bass, middle and low frequencies, Dynaudio is essentially saying ‘we’ve got it right, but if your room is excessively bright or dark, try this.’ It’s not a huge difference, but a nice way of adding tonal variation while limiting the need to get too tweaky. The rest of the rear panel switches include placement options for Free, Wall and Corner, and the rule of thumb here is 50cm. Within 50cm of a wall? Go with Wall. Within 50cm of a corner? Go with Corner. Beyond that, keep it on Free. There are three more placement considerations labelled Anechoic, Desk and Soffit. Desk is self-explanatory. Anechoic isn’t a setting for anechoic chambers but for well-treated studios, and Soffit is for when the speaker is mounted in a wall cavity. UP FRONT ABOUT SOUND
The Core series speakers all use bass reflex enclosures with front-facing ports that extend almost the width of the cabinet (with the exception AT 43
of the sub, which uses a sealed enclosure). The reflex portion of the Core 7 seems to work well. Low end is always problematic in less than ideal environments, but I couldn’t get them to show any detectable port noise. As you’d expect of a bass reflex design, taking a peek behind the driver assembly doesn’t show a whole lot. There’s a slight divide in the cabinet between the woofer and tweeter sections with some high-density foam lining the cavity. Looking inside most speaker boxes is fairly underwhelming unless you’re privy to the countless hours of modelling to get to that point. Beauty in simplicity. I’ve come to appreciate front-ported designs more as time goes on, especially for untreated or moderately-treated spaces, mainly because you know what you’re getting. Rear porting seems to change the response of the speaker more depending on the environment it’s placed in – changing anything about the speaker’s position or the surrounding surfaces can significantly alter the response of the system. They’re also just less flexible in placement. Front-ported monitors like the Core series are very happy to sit against a wall, which is handy in smaller spaces (accompanying acoustic issues aside). Also, with the modelling available to designers these days, low end AT 44
Dynaudio has always done a great job of presenting the critical vocal range
extension can be achieved without having too much of a resonant bump. The Core 7’s frequency response goes from 45Hz to 27kHz (±3dB). For those wondering how low that goes in terms of music, using a sub synth I found that the tone starts to drop off at G, two octaves below Middle C. The F# is still relatively prominent, and gradually tapers off until you hear no low tone at C (three octaves down). A lot of smaller speakers can reach a similar low frequency but it can sound artificially supported by the port,
creating high points and deader spots throughout that last octave. The low end response will also usually fall off a cliff beyond that point. The Core 7 holds it together well, right down to where it tails off gently. It seems quite accurate and happy to live down low. UNDER COVER OF DOMENESS
The dome tweeter is an impressive beast, offering great accuracy and an effortless natural tone. Dynaudio has always done a great job of presenting the critical vocal range, and the crossovers never get in the way. On the Core 7 the crossover point is pushed up to around 2.3kHz. There’s no masking of high frequencies, delay tails are all there, and discerning between layered percussion is made simple. CONCLUSION
The Core 7’s feel great to listen to and mix on. The imaging is spot-on, there’s a super wide sweet spot, and they’re really consistent off-axis. There’s a bit of competition around this price point but, rest assured, you can’t get this level of sound quality at the moment without leaping into that price territory. If you want to hear the reality of your mix, there’s a cost to getting that detail.
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antelopeaudio.com AT 45
REGULARS
LAST WORD Indistinguishable From Magic Column: Greg Simmons
In the early ‘60s Arthur C. Clarke wrote: “Any sufficiently advanced technology is indistinguishable from magic.” Thirty years later I was baby-sitting a fully-configured CEDAR (Computer Enhanced Digital Audio Restoration) system; a great opportunity because my work involved restoring audio from tape, vinyl and 78rpm discs. Pre-dating iZotope’s RX and native processing by at least a decade, CEDAR was a PC-based system loaded with DSP chips that offered real-time de-noising, de-clicking, decrackling, etc. It was amazing, and I wondered how to recreate it using the audio building blocks I was familiar with: amplifiers, filters and dynamics processing. I didn’t even know where to start. It was, essentially, indistinguishable from magic. I asked CEDAR’s Gordon Reid to explain how it works in terms of the audio building blocks I was familiar with. After some failed attempts, he said, “I can’t explain it in those terms because CEDAR doesn’t ‘see’ an audio signal that way. It just sees information; it looks for patterns in that information and responds to them.” It was my first experience of ‘post-analogue lateral thinking’... Fast forward to early 2017. I’m poring over a thread that’s unjustly ridiculing the latest field recorders. Supposedly well-informed professionals were dismissing them outright because the limiters were implemented in the digital domain, meaning they were after the converters and presumably too late to prevent overloading. “Ridiculous!” they scoffed. I’d been using these newer-generation field recorders with no limiter problems, and felt compelled to wade in with a dose of ‘post-analogue lateral thinking’. OUTSTANDING IN THE FIELD
A single channel of a field recorder consists of a mic preamp, a limiter and an AD converter. In a contemporary field recorder these components form an integrated system with no user-adjustable level controls between them. Therefore, with no limiting taking place, monitoring the converter’s output is as valid as monitoring the preamp’s output. So far, so good... There are numerous ways to make the digital limiter work, but I’m going to focus on the one I find the most interesting. In contemporary field
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recorders the preamp’s gain is digitally controlled by a rotary encoder or up/down buttons. The digital limiter monitors the output of the converter but applies the gain reduction directly to the digitally-controlled mic preamp. It protects the converter from clipping without any extra processing in the signal path. It is superior to the analogue equivalent in many ways, with one caveat: latency. If the internal processing is fast enough to match or better the attack time of an analogue limiter, it’s a superior solution and another good example of ‘post-analogue lateral thinking’. For years, conventional thinking about AD conversion said there was no point going beyond 24-bits because we couldn’t make a converter that delivered 24-bit performance in terms of dynamic range. The rule of thumb for the dynamic range of a linear PCM system is simple: 6dB-per-bit. Not so long ago we were struggling to get 20-bit performance out of a 24-bit converter. At 6dB per bit a converter with 20-bit performance offers 120dB of dynamic range, which is less than a typical condenser microphone (e.g. 125dB for a Neumann KM184). One of the goals for manufacturers whose products are intended for situations where you only get one chance to record it (e.g. live performances, location recording) is to make a converter with a dynamic range that exceeds that of the microphones. It would be one less bottleneck in the signal path, and bring us closer to a situation where setting mic gain before recording will be an option rather than a necessity. GAIN FIRING RANGE
In this issue, Stephan Schutze reviews Zoom’s F6 field recorder which, like Sound Device’s MixPre II series, offers the ability to record in ‘32-bit float’ format. It’s a different format to linear PCM so we can’t apply the 6dB-per-bit rule, but the 32-bit float format offers huge dynamic range – way more than a microphone could ever deliver. To achieve this they use a technique known as ‘gain ranging’. Instead of using one preamp and one converter on each mic input, they use two. Both preamps have fixed gain. One preamp/converter combo is optimised for low level signals, the other for high level signals. The outputs of both converters are combined through DSP to create
a 32-bit floating point signal that is essentially impossible for any contemporary microphone to drive into clipping. Gain-ranging isn’t new — you’ll find it in the AES42 digital microphones from a decade ago — but it’s another example of ‘post-analogue lateral thinking’. The F6 offers recording in 24-bit or 16-bit linear PCM format, along with the option of recording the 32-bit float signal. I’m willing to bet that the linear PCM signal is derived digitally from the 32-bit float signal, and that the knob used to control the recording gain is placed after the converters — a gamble backed by the fact that it affects the recorded level of the linear PCM signals but not the 32-bit float signal. Whatever the case, if you’re recording in 32-bit float mode from a microphone source there is no need to worry about gain anyway because you cannot overload the preamps or converters. POST ANALOGUE WORLD
Marcel Gnauk of Free To Use Sounds (see ‘Making Free Pay’) recently shared an F6 recording of a passing train, along with screen dumps of waveforms. The whistle blows and drives the 24-bit signal well into clipping. Thankfully, he was also capturing the 32-bit float signal and was able to recover the recording. It gets a bit gritty at the peak when the mic’s diaphragm reaches its maximum SPL, but that has nothing to do with the F6. Yet another example of ‘post-analogue lateral thinking’. So where is this leading? Zoom’s tiny F6 contains six ‘gain-free’ inputs. Imagine eight of them in a desktop interface, perfect for the nontechnical recording musician. Imagine 24 in a rack-mounting interface, ideal for recording live performances. Worrying about recording levels, pad switches and limiters will be a thing of the past. For those who don’t understand the technology behind this ‘post-analogue lateral thinking’, it will be indistinguishable from magic. Greg Simmons is the Founding Editor of AudioTechnology magazine, and looks forward to a time when recording musicians will look at an old interface and ask, “What does that knob marked ‘gain’ do?”
The F6. A new era in recording 32-Bit Float Recording: Capture The Calm & The Storm In the field, fluctuating dynamics can prove challenging. Loud sounds can cause clipping and quiet sounds can get lost in the noise. The F6 uses 32-bit float recording with dual A/D converters to capture both explosive and subtle sounds at full audio quality – without ever adjusting gain!
Mic Preamps: 6 pro-grade pres feature a super-low noise floor (-127dBu EIN) and up to 75dB of gain.
Precision Time Code: On or off, the F6’s Temperature Compensated Crystal Oscillator (TCXO) generates time code at 0.2ppm accuracy.
Advanced Look-Ahead Hybrid Limiters: By adding a 1ms delay, the limiters ‘look ahead’ anticipating clipping before it’s recorded.
Zoom AutoMix: lets you keep your eyes on the action by automatically adjusting the levels of your mix to reduce ambient sound.
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REIN V EN T ING T HE S TAGE K E Y B OARD
Designed for gigging keyboardists, the YC61 features a brand new Virtual Circuitry Modelling (VCM) Organ engine with physical drawbars, extensive real time control and authentic Acoustic/Electric Piano and FM synth sounds. AT 48
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