AudioTechnology App Issue 34

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

Regular Contributors Martin Walker Paul Tingen Guy Harrison Greg Walker Greg Simmons Blair Joscelyne Mark Woods Chris Braun Robert Clark Andrew Bencina Ewan McDonald Cover Image: Cameron Crone

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Daniel Howard daniel@alchemedia.com.au

Advertising Philip Spencer philip@alchemedia.com.au Accounts Jaedd Asthana jaedd@alchemedia.com.au Subscriptions Miriam Mulcahy mim@alchemedia.com.au

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

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All material in this magazine is copyright Š 2016 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. 30/11/2016.

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COVER STORY

Inside Butch Vig’s Home Studio

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ISSUE 34 CONTENTS

30

Reorchestrating A Beatles Show

Sennheiser’s Slice of Virtual 26 Action

How Do You Measure Up? That Difficult Phase

Arturia V Collection 5 AT 6

Rufus Wainwright Recomposes Shakespeare

20

Livemix Personal Monitoring System

44

34

48

Last Word with ex-Lexicon Keyholder, Michael Carnes

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GENERAL NEWS

ROLI BUILDS BLOCKS Roli has moved on from squishy keyboards to highly manipulatable touch controllers called Blocks.

The new palm-sized product sports a touch and pressure sensitive surface that lets you shape music through presses, glides, and other gestures.

Like a lot of other music technology products these days, Blocks is designed to ‘democratise’ music production — i.e. you don’t have to be a genius to create good-sounding tracks. Pitched as a modular music studio, the emphasis is on ‘shaping sound’ and taking it with you anywhere responding musically to natural gestures and pressure on the

playing surface. Blocks is a controller for Roli’s app Noise which is currently available for iOS only. The Lightpad Block is the centre of the Blocks system with the glowing pressure-responsive surface.

The Live Block and Loop Block let you record your beats, play them back, switch scales, play chords and arpeggios, and more. Multiple units can be magnetically attached together to form a larger playing area.

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

KLARK TEKNIK 1176 CLONE Well, this is weird. Of all the brands that’d be inclined to clone the classic 1176, Klark Teknik doesn’t spring to mind as the most likely. The company’s new 1176-KT FET-style compressor pays homage to Urei’s respected classic, albeit with an ‘entirely modernised’ signal path that uses Midas input and output transformers. Ratios available are 4:1, 8:1, 12:1 and 20:1, and of course the ‘all-buttons-in’ mode for that super-squished 1176 punch. Controls are an identical match to the

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original, nice and simple. In addition to the ratio buttons you get attack and release knobs, input and outputs knobs, and a power switch. A vintage-style VU meter can display gain reduction or output level based on which button is selected to its right.

The 1176-KT is housed in a 2U rack enclosure. The unit has Neutrik connectors and a universal power supply with automatic voltage sensing.

National Audio Systems: (03) 8756 2600 or sales@nationalaudio.com.au

NOVATION CIRCUIT v1.4 The latest update to Novation’s little groove box Circuit introduces several user-requested features like Sample Import and Sample Flip, Components suite of browser-based utilities, and more. A new pattern-length parameter for the drum tracks enables polyrhythmic drum sequences, where previously the drum patterns were locked to 16 steps. Also, it’s now possible to switch each track’s patterns on the fly so you can create fast-changing

fills and pattern-change effects. You can change the colour of the sessions from the unit itself (instead of from the Components software), and there are new setup options to optimise Circuit’s interoperation with other MIDI hardware. Lastly, you get newly-created samples, synth patches and sessions for even more inspiration.

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

FREEDMAN ELECTRONICS BUYS SOUNDFIELD Rode’s umbrella company Freedman Electronics Group has taken territory in the immersive audio world by purchasing the surround sound recording marque SoundField from TSL Products. SoundField invented 360° surround sound – or ambisonic – audio capture technology in 1978, and the range now includes ambisonic microphones, and systems/ apps for broadcast, music and location recording.

Peter Freedman, founder and MD, says, “Adding a product line like SoundField – with its unique surround sound microphones and applications – to our portfolio is a real step for the Freedman

Electronics Group.” Chris Exelby, Managing Director, TSL Products: “It is fitting that SoundField has now found a new home with one of the world’s most innovative microphone manufacturers, where it can be developed still further, while we at TSL Products focus on our core broadcast business and invest further in the crucial R&D that will lead to the next generation of broadcast control, monitoring and power management solutions.”

Rode: (02) 9648 5855 or info@rode.com

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LIVE NEWS NEW iPAD APPS FOR DLIVE Allen & Heath has launched two iPad apps to heighten your dLive digital mixing experience. dLive MixPad gives you comprehensive remote control of a live mix, while dLive OneMix focuses on personal monitoring. MixPad offers extensive control capability over channel processing, including filters, gates, parametric and graphic EQ, compressors and input/output delays, and gives you instant access to any of the mixer’s channel faders and mutes, DCA faders and mutes, pan, aux sends and assignments, as well as mic-pre control and full metering.

It also includes a real time analyser, the facility to name and colour channel strips, custom layers with drag ’n’ drop strip setup, and channel PFL control for monitoring. dLive OneMix is a version of the MixPad app tailored for personal monitoring, so it locks control to a single aux monitor mix. Multiple iPads can be set up by an Admin user to give artists a custom set of controls.

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

AMBER PARTNERS WITH JTS Amber Technology has signed a deal with JTS Professional Co. Ltd to be its official distributor across Australia and New Zealand from January 2017. Established in Taiwan in 1982, JTS has over 300 staff in multiple departments, The company has gained a name for producing quality microphones (both wired and wireless), conference and interpretation systems and accessories. With Amber Technology on board, the JTS brand will be represented to the ANZ pro audio, broadcast,

musical instruments, retail and commercial AV installation markets. Peter Amos, Amber Technology MD, welcomed JTS to the Amber Technology fold. “We are excited to be able to represent such a fast-growing and innovative brand in Australia and New Zealand. The JTS products will find application in the various specialised markets that Amber Technology services.”

Amber Technology: 1800 251 367 or sales@ambertech.com.au

SENNHEISER DIGITAL 6000 SERIES Sennheiser’s new Digital 6000 Series wireless microphone system is designed to be a rock solid choice for strenuous live applications. The product family includes a two-channel receiver in two different versions, a bodypack and handheld transmitter, and a rack-mount 19-inch charging unit. The digital two-channel receiver works across a switching bandwidth of 244MHz (470 to 714MHz) and features true bit diversity. The system latency is 3ms. The Digital 6000 transmitter uses the same rechargeable accupacks as Sennheiser’s

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Digital 9000 units. The SK 6000 bodypack works for wireless instruments like guitar and bass, or you can use it with clip-on mics like the MKE 1 or cardioid MKE 40. The receivers are fitted with a Link Quality Indicator so you’re warned of any issues before dropouts occur. The L 6000 charger handles up to four charging modules. Coloured LEDs display charging status on the front panel.

Sennheiser: (02) 9910 6700 or sales@sennheiser.com.au

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SOFTWARE NEWS

FABFILTER PRO R FabFilter’s new Pro R is a high-end algorithmic reverb plug designed to reproduce a very natural sound. An intuitive addition is the ‘Space’ control that switches through dozens of room models with automatic decay time adjustment, and the Decay Rate EQ that lets you adjust the decay time over the frequency spectrum. The other controls are as you’d expect to find on a reverb plug, albeit named with non-technical terms — Brightness, Character, Distance, Stereo Width, etc. The room models range

from small ambiences and rooms all the way to large concert halls and huge cathedrals. FabFilter has developed the algorithms to ensure the reverb tails sit nicely in a mix, without introducing colouration, density or phase problems. We’ve come to expect amazing GUIs from FabFilter, and Pro R delivers on that front too. The interactive reverb display shows Decay Rate EQ and Post EQ curves.

Full Screen mode offers a larger analyser of these two functions.

FL STUDIO 12.4 OUT NOW The latest version of FL Studio has been released

by Image Line. The update brings with it two new plug-ins. One is an emulation of the Roland TB303 Bassline synthesizer called Transistor Bass, and the other is FL Studio Mobile, a companion to the Android, iOS, and Windows App. You can use this to bring mobile projects into FL Studio and take them to the next level. Updated plug-ins include FPC,

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Harmless, Newtone, Ogun, Parametric EQ 2 (now with a fully resizable vectorial UI), Patcher, Razer Chroma, Slicex, Vocodex, ZGameEditor Visualiser.

Four editions are available depending on which package suits you best. The All Plug-ins version goes for US$899. The Mac version is still in the works and you can track progress at the Image Line website.

IK MULTIMEDIA MODO BASS IK Multimedia shows some love to bassists with its new Modo Bass electric bass virtual instrument for PC/Mac. 12 physically modelled electric basses form the foundation of Modo Bass — and you’ll recognise most of them. There’s the ’60s P-Bass, ’70s P-Bass, Fender Jazz Bass, MusicMan StingRay, Gibson EB-0, Rickenbacker 4003, Hofner Violin Bass, Ibanez Soundgear, and more. Even better is that every aspect is customisable; things like playing style, strings, pickups, pickup placement, electronics, tone settings, action, the list goes on.

The three playing styles are ‘pluck’, ‘slap’, or ‘pick’, and IK has spent the time to make sure you’re not limited in tone-crafting options. You can control the force applied to the strings, the position of the hand, the direction of the stroke, the impact of the thumb slap, and the finger pull of the string. Modo Bass also includes 20 pickup options, seven stomp boxes, two bass amps (derived from AmpliTube), and two cabinets.

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

STRUMMED ACOUSTIC 2 Native Instruments Strummed Acoustic 2 is out now, and it joins the Session Guitarist series to complement the original Strummed Acoustic instrument. The new version brings samples of two vintage acoustic guitars — a six-string Martin and 12-string Guild — plus more strummed patterns and new performance controls. Strummed Acoustic 2 runs in Kontakt and the free Kontakt 5 Player.

There’s a nice contrast between the two guitars.

The Martin produces the warm, mellow tones of a mahogany body from an instrument that was

originally built in 1934. The jumbo 12-string Guild, produced in the late 1960s, delivers a more full, bright and articulate sound. The new ‘Separate Bass’ feature lets you add custom bass lines or add extra low notes to chords to create slash chords.

And with Tap Rhythm Finder, you can dictate the groove of a strum pattern by tapping it into your keyboard. Strummed Acoustic 2 adds over 50% more content than the original.

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

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FEATURE

INSIDE BUTCH VIG’S HOME STUDIO Butch Vig gives AT a sneak peek inside his ‘glorified bedroom studio’ Grunge is Dead where Garbage wrote and recorded half their latest album.

Story: Mark Davie Butch Vig Portraits: Cameron Crone

Artist: Garbage Album: Strange Little Birds

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Could there be a more credible flagwaver for the legitimacy of home studios than Butch Vig? This is a guy who owned a commercial studio for 30 years and recorded Nirvana’s Nevermind at Sound City. Now he’s making records in Dave Grohl’s garage (ignore the API console — little picture, people!) and records drums in the den of his Silverlake, LA home.

“It’s a glorified bedroom studio,” says Butch, describing the space he’s dubbed Grunge is Dead.

Sure it has windows overlooking Silverlake and a fairly big Pro Tools rig. “But there’s really not much soundproofing, just a few baffles on the walls,” continued Butch, pointing out what’s in his room.

“I’ve got a couple of small guitar amps, my drum kit is here, there’s a piano. It’s a good writing room. We wrote and recorded a lot of the new Garbage album Strange Little Birds here, but we finished it all in a proper studio.”

Oh, there it is; a proper studio. He’s talking about Billy Bush’s place, Red Razor Sounds. Billy has been engineering Garbage records since their second album, and has been married to lead singer, Shirley Manson, since 2010. Red Razor is like a home away from home for Garbage, and only five minutes from Butch’s place, in Atwater.

Part of the reason Butch could move away from his own commercial studio — Smart Studios in Madison, Wisconsin — over 12 years ago, was because of the variety of studios on hand in LA. “It’s easy to work here, there are so many studios and so many people I know in the music business,” he explained.

The actual breakdown of recording done at Butch’s home versus Bush’s studio is unclear. Both of them quoted about 60% of the record done at their place; let’s call it an even 50/50.

It’s funny to think of a rock icon having to work around the schedules of his neighbours, but that’s exactly what Butch does. He records all his drums at home during a four-hour window in the afternoon when “I can make as much racket as I want,” he said. “Around dinner time I have to stop.

My studio has no soundproofing, it’s just a den, a rectangular room with dry wall that doesn’t even sound that good. You can hear the drums through the whole valley here. I’ll run into my neighbours when I’m walking my dog and they’ll say, ‘I heard you playing drums yesterday, what was the song you were working on?’ I really like my neighbours, so I keep it to that window.”

HOME STUDIO STYLE

While Butch’s home studio might be relatively untreated, the gear on hand isn’t exactly cobbled together. His “basic setup” includes API and Helios preamps, Neve and Harrison EQs, Chandler EQs and preamps, and his ‘secret weapon’ Roger Mayer RM58 solid state compressor, which “kind of f**ks the drums up.” He’s got a couple of guitar amps, Fender and Matchless, as well as a Line6 Helix and Kemper Profiler hooked up directly to his Pro Tools rig, his drums, soft synth controller keyboards and the upright piano that Duke likes to play. On the mic side, he has a 1959 Telefunken ELA M250 and a Cathedral Pipes U47-style tube

mic. However, “Shirley just likes to grab a Shure SM58 handheld mic,” said Butch.

Butch has Grunge is Dead for the same reason most musicians have a home studio; to have a place where he can work on his own music at leisure, and in the way he likes working. “It’s wired so I can open a session and record anything instantly,” he said. There’s no distinction for Garbage between a demo and final master, said Butch. If grabbing an SM58 and singing from the couch gets the vibe Shirley’s after, then he could care less about which mic she’s using. “Once we start recording, that could turn into a master.

Sometimes we get very meticulous in trying to get a sound that fits the track. Other times we don’t care. We just turn a mic on that’s nearby and record it, even if it’s 10-feet away. We just want to get the idea down. That’s been a tradition with Garbage since the first record.”

Garbage aren’t under any pressure to make a Top 40 hit. “Even if we wrote one, it wouldn’t get played anyway,” stated Butch. “Acknowledging that frees you up to do whatever you want.” Rather than bending over trying to contend with the latest pop generation, Garbage decided to return to the experimental roots of their debut, where nothing was off limits and anything could be sampled, recut and reimagined. It also meant an iterative approach to production that works better when there are no studio overheads. “We just started hanging out and talking, telling jokes, sometimes we’d listen to other music or watch a film,” said Butch. “Then we’d just start jamming. If it happens, great! If not, well let’s come in tomorrow and try again.”

NEVERMIND, JUST HAVE FUN

While this record may not be the cultural milestone that Nevermind was, or even the first couple of Garbage records, Strange Little Birds showcases the creativity of a true sonic mastermind in his element, surrounded by his most long-serving collaborators.

There isn’t one mode or style to Garbage on the album, while the song Empty has all the trademarks of a Garbage hit — slammed drums with complementary programming, eighth-note rolls, layered guitars that still punch through, big wall of sound chorus and a simple, catchy melody — other tunes feel worlds away.

Album opener, Sometimes, alternates between modulated, tremolo noise and orchestration with a dark cinematic quality to it. Shirley’s voice, on the other hand, is starkly dry. It’s a startling album opener, and even more ‘in your face’ than Empty’s big rock chorus. “We wanted that juxtaposition because it felt nerve racking,” explained Butch. “It made your brain pay attention to it. We could have put it all in the same reverb space, but we wanted Shirley’s voice to have this otherworldly quality and be really confrontational, confessional and vulnerable. The way to do that is take the effects off, push it way up in the listener’s face. Because then you can’t escape it.”

Empty, like a lot of songs on the record, started out as a jam to a programmed beat, but it didn’t feel right until they humanised it. “We had the

chorus so I put down a quick kick and snare pattern in Battery, but it felt stiff because it was just a constant tempo,” said Butch. “A couple of days later, Duke and I played the song without a click track. We did three or four takes and picked out bits that sounded good. Then I went into the Pro Tools session and looked at the tempo of the verse, 109bpm, then I’d look at the tempo of the chorus which was 111bpm, and noticed I was speeding the drum fills up to 115bpm. It’s okay, because it sounds good. Then I made a variable tempo map based on our jam without a click. I reprogrammed the kick and snare to the new tempo grid and went back and re-recorded the drums over that. It was just a process to get it where the ebb and flow of the tempo felt natural.”

I’ll run into my neighbours when I’m walking my dog and they’ll say, ‘I heard you playing drums yesterday, what was the song you were working on?

BUILDING WALLS OF SOUND

The chorus of Empty features the classic Garbage wall of sound, which Butch puts down to being “very layered. There’s a lot of guitars with a specific tuning. We used a drop D, but tuned the high E string up a step to play the inverted note at the top of the chord. You get an orchestral sound along with the keyboard parts we played.”

At other times, the heaviness of the guitars came from dynamics, juxtaposing quieter parts against a sudden onrush of sound. While the verse and chorus for Empty came quickly, the bridge took three weeks on and off at Red Razor to get the power they were after. Butch: “We tried stuff that pushed harder and took off, and a couple of different time signatures, but it got really big so we decided to pull it back and drop it out. When Shirley drops down to that quiet vocal, the guitars have to leap out and really exaggerate the dynamic.

Billy has about 100 stomp boxes, so we spent a lot of time trying to find the right amount of fuzz and how far we could push it. It’s a tendency we have with dynamics like that, to try and really oversaturate them.”

Magnetized is another song that elevates into a wall of sound, but wasn’t as simple as pulling up a Garbage preset. It was by far the longest birth of any song on the record, taking them a year to complete. “Steve’s initial demo was way slower and sounded like the Jesus & Mary Chain on drugs,” recalled Butch. “Shirley heard it and immediately sang, ‘I’m Magnetized’. This glorious vocal melody that sounded like a Roy Orbison melody.

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GARBAGE COMPRESSION One of the not-so-secret pieces of outboard that has helped define Garbage’s sound is the Roger Mayer RM58 compressor. Butch and Billy swear by it for drums.

Butch: “If you listen to the drums on the end of Amends, they’re hammered through the Roger Mayer, which blows them out in a cool way. Whenever the drums get really loud, like on Empty, it’s probably mixed in 50% on one of the buses.

“I found it when I was producing Sonic Youth’s album Dirty in New York City. The engineer, John Siket, knew this gear broker who had a bunch of weird gear out in Brooklyn. I went out to his pad and he said, ‘You should try out this compressor, man.’ It was a $500 solid state compressor. I took it to the studio, plugged it in, and to me, it made everything sound like early The Who records.

It had this wild, but cool, crunchy out of control sound. So I bought it. I’ve used it on so many records, but it’s got a specific sound, so you either have to embrace it or not.”

Billy: “Roger Mayer made these compressors in the late ’60s when he was making a bunch of console parts around New York. They’re super fast solid state compressors that admittedly don’t really sound great. I called Roger Mayer up about it and he said, ‘Why would you want to use that?’ When you put it in the most extreme settings, it’s got the most glorious drum sound.

We scoured the planet for these things and we’ve got three of them now, Butch has got one and I’ve got two. They’re beat to hell, the VU meters don’t work, and it only ever works when everything’s cranked to 10. If you run it as a parallel compressor on the kick, snare and toms, and a little bit of room mic, it creates this sound I haven’t heard on anything else.”

“We tried it a couple of different ways: hyped and aggressive like how we’d play it onstage, and with a string arrangement. One day we plugged in a stomp box called a Glamour Box, which just makes noise. We’ve done that on previous Garbage records, when in doubt just record take after take of noise and see what you come up with.

“We started getting all these weird tones out of the Glamour Box that I cut into pieces to get more of a musical sense to it. Then when it gets to the chorus, we came up with this symphonic sound with all these fuzzed out, hyper-sounding guitars, and a lot of keyboards. It became a lot more orchestral and cinematic and a lot less rock ’n’ roll, which made sense.”

ARRANGEMENT BY ADD

Even if you’re not a Garbage fan, one thing you can say about their songs is they’re rarely boring.

There’s always a new sonic element to sink your teeth into, sometimes every couple of bars. Butch: “All four of us have some ADD-ish tendencies.

We’re always saying, ‘we need something new here to keep the song building’. That goes back to the very first Garbage record where we did tons of layering then defined the arrangements in the mix AT 16

Butch’s clear acrylic dw kit is miked up with a pair of pose-able head Blue Hummingbird condensers overhead, Josephson e22S side address cardioid on the toms, a Shure SM57 on snare top and a Telefunken dynamic underneath. Room treatment courtesy of platinum and gold records, they really give the tracks some weight.

Trying to make an SM58 sound like Telefunken ELA M251 has its challenges, but I managed to get away with it

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Billy: “At the top of my two pedalboards are Audio Kitchen The Big Trees pedals, which are these great preamp and distortion boxes. A lot of time we’ll take a soft synth, go out through The Big Trees then back in, and it will make it sound more real.”

by muting and bringing other parts in.”

That proclivity for newness can produce some pretty dense arrangements, but occasionally keeping that ADD tendency in check is exactly what a song needs. Even Though Our Love Is Doomed was a track that went through stages of being built up before being stripped back to its original version. Butch wrote the song and recorded a few demo versions at home, none of which he liked. A month after he’d mentally shelved it, Shirley recommended he try record something simpler. Butch: “The next day, I came down to my home studio. All I really had was the melody line and some words for the chorus. I thought I needed to have a verse, so I picked up the bass and started playing a line that became the core of the song.” 45 minutes later he had a whole song; music, lyrics, the whole lot.

It was pretty spare, but that was part of the appeal.

“Shirley loved it,” said Butch. “She heard it once and sang the vocal first time. Being a demo, we thought we’d try add stuff to it — bigger drums, and Duke and Steve tried a few parts too. But we didn’t like them and the demo is pretty close to the final version. There was something about it that had a spontaneity or intensity to it because it feels like it’s holding back.”

Another way Butch keeps the arrangement from getting overcrowded while maintaining interest in the song is by using the stereo field to create dynamic and change. Butch: “Initially, it’s good to have a space where your brain hears the song and settles in. As the song goes on I like to hear things that move around a little, spatially. Maybe the second verse doesn’t need a new part, maybe we just need some panning or a filter. It’s the remix mentality from the first record; what can we do with the same parts?

“I never listen to the panning in headphones until we’re pretty much done with the mix. The nature of how headphones isolate things to your left or right ear means you hear panned elements much sharper and you’re more conservative with how you build effects. If you’re sitting in a room with stereo speakers they’re still blended in with the AT 18

sound of the room, so you can push up stuff crazy loud and do crazy things with them. Later, when you put the headphones on, you think, ‘shit, that’s really loud in the left ear!’

“There are a lot of different ways you can pan things in a mix, whether it’s organically by recording it stereo, or using outboard or plug-ins to put it in that space. There were so many layered parts on the first Garbage record that were panned left and right. We’d pan a dry sound in the verse to the right and have another part in the left channel that we’d filter really tiny so you could barely hear it. It was just trying to keep your brain moving, so every time a new section of the song came up you’d get some new ear candy.”

MIX ’N’ MATCH

Over time, Garbage mixing duties have moved from a partnership between Butch and Billy, to Billy primarily mixing with Butch helping tweak it towards the end. They’ve developed a trust that allows Butch to stay in the songwriter/band member mode for longer. Billy: “My goal with a Garbage record is to have something that sounds immediate, but also on subsequent listens, discover layers you haven’t heard before. The mix also has to channel the emotion attached to the song as much as possible. I ask myself the question, ‘Am I getting the emotional response off it I should?’”

Once everything is laid out in the session, with all his routing intact. Billy always starts with a combination of the groove and the vocal. He’ll start with the loudest section of the song and get it to be as loud as possible, then work the dynamics of the rest of the mix around that. “A lot of it is finding out what are the most important moments that need to come across,” he said. “Getting the chorus to hit really hard is about getting all the melodic and harmonic information working. I find how to approach the frequency range of all the elements of the song, which allows you to have multiple layers of drums and kick drums, synth basses and guitar bass and all the guitar tracks. We try to figure out where it’s going to fit ahead of time while we’re recording.

Rather than just having one super saturated guitar

that takes up all the frequencies, we figure out combinations that add weight in different areas.”

Because Garbage demo recordings can filter right through to the final master, there are times when Billy has to make things fit. Billy: “The chorus vocal of Magnetized is the one Shirley originally tracked with an SM58 on Butch’s couch.

It was in a different key and tempo to the final track, but it had the right feel to it. Besides having to make it fit into the structure and key of the song, I also had to tonally match it with the rest of the vocals recorded with my Telefunken ELA M251.

Trying to make an SM58 sound like 251 has its challenges, but I managed to get away with it.

“Conversely, on Even Though Our Love Is Doomed Shirley wanted to take one line of one verse she sung in my studio and try make the Telefunken sound like a 58! I would sometimes slide over into Logic and use the EQ Match feature or just use radical EQ and compression. I had to try and find the elements that were working in the mix for the rest of the song and make the microphone fit into that.”

Sometimes it’s not just one part Billy has to manipulate, but an entire section.

Billy: “The band want to get ideas down quickly, but invariably the tempo and key almost always changes. They write a lot of songs from the chorus or the bridge, then write a verse much later. It might make the chorus uncomfortable to sing, so they change it. Still, they might really like the tone of the original take so we might drop it down a whole step and speed it up 8bpm. It’ll be a combination of Serrato or Melodyne, and sometimes cutting it by hand to get it to the proper tempo. It’s not to say they’re lazy, it’s about keeping the magic.

“The most we time stretched something was 18bpm. They’re always relaxed and super chilled when they write, so it tends to be a much slower tempo. I’m always speeding it up. Sometimes we’ll already be in mix mode and someone will suggest one bpm faster, then I’ll have to Elastic Audio the entire song!

“A lot of that stuff used to be a real nightmare on tape. It’s a lot faster now, and allows the band to stay in a more creative place for longer because they have a computer to create 65,000 takes on. It used to be, ‘Can you speed this up 2bpm?’ And I’d say, ‘Yeah, okay, go to lunch and come back in an hour and see how it turned out.’”

Butch: “A lot of Garbage songs come together in the final mix. We have a tendency to record a lot more ideas than necessary and not discard them.

Once we start an idea for a song — whether it’s a jam or a demo — we just keep adding. Most of the songs are defined by what we take out.

“It can be a stressful experience in the mix. It usually takes us two or three days to mix a track because we’re often making the final decision of what the final arrangement is going to be. The four of us argue about all those things, and we don’t always agree. More often than not a Garbage song has a path it goes on, and none of us are really sure where that path is going to go until we print the final mix.”

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FEATURE

Rufus Wainwright has never been one to settle on any style, but lately he’s been obsessed with one particular artist, Shakespeare. Maestro Marcus de Vries helped Wainwright pull together a diverse tribute to the bard for his 400th birthday.

Story: Paul Tingen

Artist: Rufus Wainwright Album: Take All My Loves, 9 Shakespeare Sonnets

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As the son of folk-music luminaries Kate McGarrigle and Loudon Wainwright III, Rufus Wainwright was expected to carry on the family business. However, the last thing Wainwright junior wanted to do was pigeonhole himself as just a folk artist. Over the years he’s wilfully dived off any stylistic cliff into genres like cabaret, music hall, Broadway musical, Tin Pan Alley, jazz, Sinatra crooning, occasionally diverting into the slightly more obvious options of pop, rock, and yes, even folk. Wainwright’s musical output has been anything but static. Lately, however, he’s become obsessed with Shakespeare.

It all started out relatively innocuously with an invitation by avant-garde director Robert Wilson and the Berliner Ensemble to set a number of Shakespeare’s sonnets to music. It was for a theatrical performance called Shakespeare Sonnets which premiered in Berlin in 2009. A year later, the San Francisco Symphony jumped on board the Shakespeare train and commissioned Wainwright to orchestrate five of those same sonnets.

Getting further mileage out of the bard’s work, three of the sonnets found their way in adapted form to Wainwright’s piano/vocal album All Days Are Nights: Songs for Lulu in 2010. Five years later — following a rock ’n’ roll-influenced outing, Out Of The Game and Wainwright’s first opera, Prima Donna, released on the hyperprestigious classical music label, Deutsche Grammophon — Wainwright released Take All My Loves, 9 Shakespeare Sonnets, a full-blown ode to Shakespeare that coincided perfectly with the 400th anniversary of the bard’s birthday.

Sequenced as one whole, multi-faceted work, Take All My Loves compromises 16 tracks. Five of which are the same works he orchestrated for the San Francisco Symphony, but performed by the BBC Symphony Orchestra and sung by soprano Anna Prohaska with Andrew Keener producing.

For the remaining 11 — where it gets even more interesting — Wainwright worked with Marius de Vries, who — like Wainwright — has made the transition from the world of pop and rock to classical music maestro. The 11 tracks include six brief sonnet recitations by the likes of Carrie Fisher, Helena Bonham-Carter, William Shatner and others, with backdrops provided by de Vries and his son Ben. One major non-orchestral, German language track was recorded in Berlin while de Vries was busy in LA. The remaining four songs are the title track, which is propelled by electronic drums and a heavy bass line, and has minimalist piano and scores of vocals; Unperfect Actor, a Velvet Underground-like, heavy guitar outing; an elegant pop song sung by Florence Welch called When in Disgrace with Fortune and Men’s Eyes; and a delicate, moody ballad, A Woman’s Face. It sounds like a crazy mishmash, but the overall effect, including the orchestral tracks, is a knock-out.

GETTING THE PICTURE

“It’s good, isn’t it?”, enthused de Vries the moment I connected with him in his LA-based Berry Drive home studio. He carried on to explain that it was “a funny” album to make for him, because he’s

been “so deep into movie music during the last five years. This is the first non-movie project I’ve done for a while.” He ruefully acknowledged there’s more money in movie music these days than in his old metier of crafting avant-garde pop. “It’s interesting to go back into the studio without a picture in front of me,” he added. “I do it for the love of it, nowadays. The people I do music-only projects with are old friends, or they do something that’s profoundly, creatively interesting to me. Both were the case with Rufus’ Shakespeare album.”

William Shatner is my neighbour, so I just walked up the hill and asked him! We recorded Carrie Fisher in her bedroom when she had a day off from Star Wars

Take All My Loves’s blend of genres, moods, instruments and instrumental colours was “profoundly and creatively interesting” to work on, said de Vries. As were those sonnets, which have been the bane of quite a few English students. “Yes, the sonnets are hard to understand,” empathised de Vries, “but they’re easy to feel when spoken or performed. You can get the benefit of them without understanding every word. THE TRUISM IS YOU CAN’T REALLY SING POETRY, BECAUSE IT HAS ITS OWN INTERNAL RHYTHM AND MELODY, BUT RUFUS IS A PROFOUNDLY GIFTED LYRICIST, AND HE HAS FOUND WAYS IN ALL SORTS OF DIFFERENT STYLES TO MAKE EXTREMELY COMPLICATED WORDPLAY WORK WITH MUSIC. The compositions he

had come up with in this case were so deep and interesting they made my job as a producer easy.”

Wainwright made contact with him at the end of 2015, during a break in de Vries’ work on the feature movie Lalaland. According to de Vries, “We had some scores and string recordings from the Berlin performance in 2009, which we incorporated later. But as a first step, Rufus came in and played me the music on piano or guitar with a scratch vocal. I recorded that in Logic and built drums and sounds around that. He’d say, ‘I like that,’ or ‘I don’t like that.’

“We worked really quickly. The sketches for every single one of these four tracks came together

in a day, then we dressed them up with real instruments. Although we tried to settle on the tonal language, I realised the whole thing was so eclectic there was no point trying to turn it into one piece of cloth. The album was to become a journey with so many twists and turns, so we decided at a very early stage not to be frightened of that. In the end I think it’s what really makes it work.”

DEMO DRIVE TO IGLOO

This first demo stage took place with the two holed up in de Vries’ Berry Drive one-room project studio. “I work in Logic most of the time, using an Apogee Quartet and Focusrite Liquid for I/O, and KRK VXT8 monitors,” outlined de Vries. “I use my own EXS24 sample library for many of the sounds, particularly drums, which I program beat by beat.

I’m also a big fan of the U-He synths, particularly the Zebra and Ace. The Box has become very good now, it really has, but I continue to do a lot of stuff outside it. I still love my ARP 2600, my EMS VCS3, and the MiniMoog Voyager. The tactile feel I get from touching my old synths is still very important to me. My main keyboard is the 88-key Korg Kronos, because it has a lovely feel. It also has a good array of controllers in it, which allow me to tell Logic what to do. I’VE ALSO BEEN REALLY GETTING INTO THE ROLI SEABOARD GRAND, WHICH I USED ON THE TITLE TRACK OF RUFUS’ ALBUM. IT HAS REJUVENATED MY INTEREST IN PLAYING TECHNIQUE.”

In addition to de Vries’ work, additional drum and percussion programming was performed

by Eldad Guetta and Ben de Vries at his father’s Hanway Place studio in Soho, London. The next stage of the project was for de Vries and Wainwright to go and record some of those “real instruments,” which was done at Igloo Music, a large facility in LA with four main studios and several additional editing suites. Igloo is unique in that it specialises in both music and film work, and employs a whopping six full-time engineers. One of those engineers is Nicholai Baxter, whom de Vries first worked with on the film Sucker Punch.

Baxter has won three Grammy Awards, divides his time 50/50 between music and film work, and also works independently. He engineered and later mixed the tracks de Vries had been involved in.

Drummer Gary Novak, bassist Chris Chaney, and guitarist Joel Shearer were all recorded at Igloo.

“We did everything crazily fast,” recalled de Vries.

“Rufus had spent 10 years in preparation and we spent maybe just three or four weeks pulling it all together. We got some really great musicians in, and a fantastic engineer, and it was done before we knew it. We set up Igloo initially for Unperfect Actor, and started by recording Gary. Joel’s guitar parts were born out of a guitar arrangement created by Dom Bouffard in Berlin in 2009, and we also used some of Dom’s parts. The idea of this repetitive, Velvet Underground-type guitar riff on one chord was already there. Joel really went for it — his hands were bleeding by the end of that session! After that we overdubbed the bass.

“The song Unperfect Actor was our main focus,” confirmed Baxter, “and we wanted it to have a gritty, driving sound. After we had recorded the AT 21

Mixing Take All My Loves Vocals Nicholai Baxter: “In this section you can see the lead vocals at the top, entering throughout the song, and the stacks of backing vocals below. Almost all tracks in every session have the Slate Virtual Console inserted, which I use to give every track some character to start out with. Insert ‘7’ is a UAD 1176, which is the best-sounding 1176 modelling plug-in I’ve heard. Insert ‘P’ is a Fabfilter Pro-Q2, which is a great and versatile EQ. “I had trouble using a de-esser on Rufus’ vocal and ended up using clip gain automation to take out the esses. It sounded more natural, so I did the same thing on the other vocals. Insert ‘M’ is the Massey de-esser, which I tried and then turned down.

‘D’ is for Decapitator to add some fatness, grit and character to the vocals. The sends are going to delays and reverbs. I used Echoboy delays, and an outboard TC 6000 for reverb. The BVs are all sent to a bus on which I have Cytomic’s The Glue. It’s usually a master bus compressor, but I’ll use it on buses as well sometimes. Then there’s the Pro-Q2 EQ and the Brainworks V2, which I use for width. It was a natural sounding way to spread the BVs around the lead vocals, which were more in the centre. MBSR is the Massey de-esser, and TC1 is a send to my outboard TC6000.”

Mastering Engineer Eric Boulanger ensured the crucial dynamics of the album were kept in play, as you can see in the waveform of Take All My Loves. “It was a thrill for me to know the fidelity would be maintained through mastering,” said Nicholai Baxter. “Sadly the loudness wars are creeping into film work as well, with movies getting louder and limiters being used at the dubbing stage, which was never the case before. Unfortunately, it’s starting to become fatiguing to listen to feature films as well!”

MARIUS DE VRIES BIO The pursuit of music that is, as he says, “profoundly, creatively interesting” has informed de Vries’ entire career. From South-African descent he was born in London, and received his first musical education at the St. Paul’s Cathedral Choir. He later became a keyboard player and programmer and then a co-writer and producer, eventually working with well-known artists, including Massive Attack, Madonna, Björk, Robbie Robertson, U2, PJ Harvey, and others. It’s fair to say that de Vries was one of the main architects of the wave in ’90s music that took its inspiration from the latest developments in music technology.

From 2000 de Vries’ work became increasingly diverse, abandoning obvious signs of being in the music tech vanguard, and collaborating with the likes of Josh Groban, Elbow, Marc Almond, and Rufus Wainwright. He produced the singer’s second and third solo albums, Want One (2003) and Want Two (2004), and mixed Release The Stars (2007).

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De Vries first wrote movie music for Romeo+Juliet (1996), then in 2002 played a central role in making the soundtrack for the film Moulin Rouge. In 2010 he went to LA to finish off the movie score for Sucker Punch, and then found himself in San Francisco, working with George Lucas for a long time on Strange Magic. Since moving to LA, de Vries has “been involved in a constant stream of long film projects. Even as I still try to find time to make the occasional record, the synergy of music and film has become what I do.”

The truism is you can’t really sing poetry, because it has its own internal rhythm and melody, but Rufus has found ways to make extremely complicated wordplay work with music

Strings NB: “The timing edits you can see here were done by me. This track has a 50-50 blend of live and programmed strings to give it the sound that Rufus and Marius were after, so there is a lot of getting the live strings to match up with the programmed strings and gel together. MA and MB are the room mics. The third track was a weird distant room mic. Marius’ sampled string tracks are somewhere else in the session, as a stereo track. VMR is the Slate Virtual Mix rack, which I have the VCC running in to create character and vibe, and to fatten up the strings. I also use the UAD Shadow Hills compressor, which adds great character to strings. The signal then goes to the Fabfilter MB multi-band compressor to control dynamics and dial back features that are popping out too much. Then there’s some EQ and a Decapitator to add more character, grit and fatness wherever we could, and a little bit of room reverb.

The string recording was really tight sounding, so we needed to add a bit more space on it. The TC send also goes to the TC6000.”

drums, bass and guitars for it, we backed off the compression and distortion for the overdubs on the other three songs. Unperfect Actor was the hardest to get right, both in terms of recording and mixing, because it’s so cyclical and drone-like. To keep it interesting we had to get the drums and the guitars to build in just the right way. It grows and grows until it peaks and releases into this very gratifying tonal shift. That took a few takes, but we got it right at the end of the day.

“For the drums I mostly used a Neve Kelso 12-channel sidecar, which has awesome pres with lots of character that compress things on the way in. I used a rock miking setup, with many ribbon mics for a fatter, grittier sound. I had 4038s for overheads, the majority of the drum sound came from them, a Neumann M49 as a mono kit mic, two Royer 121 room mics, and an RCA 44 which I really drove through the Kelso pres and used as a trash mic. I also had a Chandler compressor and Distressors on the room mics. I had an ElectroVoice RE20 on the bass cabinet, going into an Avalon 737 and a Teletronix LA2. On the guitar cabinets I had Shure SM57s going into Neve 1073s, and then into Distressors.”

PULLING A FEW STRINGS

The final production stages on three of the four main songs was inserting the strings which had been recorded in Berlin seven years earlier, then adding vocals. “We retrofitted the old recordings to our tracks,” explained de Vries. “I had also added sampled strings, so the live strings had to fit with

those as well. The exact nature of the string sections varied from track to track, with some being larger than others. We had to do quite a bit of clever editing without using elastic audio as it changes the sound too much. In some cases we had to change the pitches of some of the notes; it’s amazing what you can do with Melodyne these days!”

The sonnet sung by Florence Welch didn’t have any strings, but de Vries recalled other challenges: “Rufus had persuaded Florence to sing the song and they found one day in London when she could.

There was some frantic back and forth across the Atlantic as to what key she wanted to sing it in. In the end we recorded the song in four different keys, with Rufus doing guide vocals that were way out of his range. He sounded like this weird baritone Rufus! Florence chose the key, and then she went in and recorded the song, only for them to realise they had set the sampling frequency wrong: they had recorded her in 48k rather than 44.1k, so she was more than a semitone out of tune. She had to go back in and sing it again, and she nailed it. At the very end of the recording process I recorded Rufus’ final vocals at my place, using my favourite black Neumann U87, going through an Avalon VT737SP, without compression. It’s my tried and tested vocal chain. With people who can really perform, the vocal chain is always very simple!”

TALENT ROUND UP

Other noteworthy production aspects of the four main songs include an epic crackle in A Woman’s Face and what sounds like telephone sounds on

Take All My Love. De Vries elaborated on the why and how: “Ha, that crackle is pretty prominent!

It’s from an old break beat from my ’90s sample collection I used to get the song going. I can’t shed my ’90s habits, in particular my Massive Attack habit! What sounds like a telephone sound on the title track is just me messing around with the ARP 2600, adding an introductory tone that seemed to work for the song. With a song like that, which is very drone-like and repetitive, you need a kind of signature sound to get people used to that language and draw it together. I also programmed the drums, with a contribution from Eldad, and played that swooping melody in the intro on the Roli keyboard. Other than Chris’s bass overdub that track is almost only Rufus and myself.”

At the end of Take All My Love, de Vries also reads the sonnet. It was part of a late decision to intersperse the main music tracks with brief sonnet recitations, and in two cases — Unperfect Actor was the other — integrate the recitations into the songs themselves. “I’m very uncomfortable hearing my own voice full frequency,” admitted de Vries. “So I used the AudioEase Speakerphone plug-in on the Bakelite preset.

“We arrived late at the spoken word idea, but it turned out to be crucial to the record, because it gives some punctuation between the many different styles. The people who did the recitations were whoever was around at the time. William Shatner is my neighbour, so I just walked up the hill and asked him! Funnily enough he wanted to do it at his office on Ventura Blvd, where there’s a lot of AT 23

traffic. I recorded him with a handheld Roland recorder, and when we got home we realised the audio was compromised. His recitation was great, so I suggested to Ben to filter it and chop it up and make some dramatic spaces and add some echo and abstract sounds. Rufus stayed with Carrie Fisher for a while, and we recorded in her bedroom when she had a day off from Star Wars. Helena Bonham-Carter is a friend of Rufus, she was recorded by Ben at my house in London. Ben and I added textures behind some of these recitations because we wanted something a bit more abstract and musical. It sounded too brutal to have just spoken word in between the music pieces.”

SONNETS IN SEQUENCE

By far the most difficult task of the entire project was sequencing then mastering 16 tracks of material that are outrageously diverse, not only in terms of musical styles and sonic colours, but also in dynamic range. Before that stage there was still the small matter of mixing the four main tracks de Vries and Wainwright had collaborated on. During the recording stage, people had worked on material in LA, London, and Montreal (where Martha Wainwright was recorded at her brother’s Mayk Music studio). Then in LA, de Vries loaded everything back into the Logic sessions for editing, comping, occasionally tuning vocals and doing rough mixes. The next stage was for him to convert the sessions to Pro Tools and send them to Nicholai Baxter at Igloo.

“I started the mixes on my own in my room, Studio D,” recalled Baxter. “Then Marius and Rufus came in and we dug into the details together. Rufus had a very specific vision for the way he wanted each track to hit you. He did not want any half measures. If a song was intended to be dissonant and aggressive, we went for it full force, and if a song was intended to be mellow, we went 100% in that direction. I used an Avid Icon D-Control during the mixes, which helps with workflow.

The custom fader banks and ability to grab knobs makes things quicker.

“There’s tons of automation in the sessions, especially in Take All My Loves and Unperfect Actor.

In Take All My Loves the vocals are becoming more AT 24

dense throughout, and it took a long time to get right. We needed to preserve our focal point while maintaining the song’s momentum. There was also a cacophony of background vocals entering and exiting. Unperfect Actor was difficult to mix, because of its shape and dissonance. While mixing I needed to have certain dissonant sections on loop for hours, and that became quite challenging. There are minor and major tonalities rubbing against each other for long stretches in that song! We also did some additional overdubs during mixing. Rufus added some vocals using a Neumann M149, and I added an acoustic guitar recorded with an AEA N8 ribbon. There wasn’t much mixing to be done on the spoken word tracks, as Rufus wanted them really raw.

“After the mixes, we spent a long time sequencing and levelling the record, and making sure we had the just right spacing between tracks.

It involved many passes of listening to the whole record all the way through then making tweaks. WE WERE ALL ADAMANT ABOUT MAINTAINING THE ENORMOUS DYNAMIC RANGE OF THE RECORD. IT WOULD NOT HAVE WORKED HAD WE CRUSHED IT. THE RECORD NEEDS TO BREATHE, SO THE LOUD SECTIONS CAN HAVE THE IMPACT THEY ARE INTENDED TO HAVE. It was a thrill for me

to work on this project for this reason alone, to know the fidelity would be maintained through mastering, which was done by Eric Boulanger.

Sadly the loudness wars are creeping into film work as well, with movies getting louder and limiters being used at the dubbing stage, which was never the case before. Unfortunately, it’s starting to become fatiguing to listen to feature films as well!”

“The mastering was fantastic on this,” agreed de Vries. “We did a lot of crossfading and Eric really helped draw the album’s many disparate elements together. The album is a significant achievement and is already finding an audience. People respond to it and spread the word. Meanwhile, Rufus is just carrying on doing what he does, as am I. I’m back on the Lalaland project, and later this year I’ll be in London working with Chrissie Hynde on an orchestral jazz dub album. For now I’m just thankful I’ve been able to contribute to another Rufus Wainwright record!”

Nick Baxter and Rufus at Igloo Studios where the album was mixed in Studio D. Baxter: “Rufus had a very specific vision for the way he wanted each track to hit you. He did not want any half measures. I used an Avid Icon D-Control during the mixes, which helps with workflow. The custom fader banks and ability to grab knobs makes things quicker.”

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AT 25

FEATURE

SENNHEISER’S SLICE OF VIRTUAL ACTION Sennheiser is preparing for a future where microphones can go anywhere, even into virtual worlds.

Report: Mark Davie

“VR could be the next 3D,” warned Simon Beesley, Sennheiser’s Manager Global Business Development for Broadcast & Media. If you haven’t kept up with screen tech in the last decade, that’s not a good thing. 3D died under a mountain of CGI action movies and 3D-‘enhanced’ detritus controlled by an unholy alliance between Hollywood and screen manufacturers looking to cash in quickly on ‘the next big thing’.

While Hollywood keeps its factory line of Avengers 3D movies trickling along, the tech industry, and many screen manufacturers have moved on to the next glittery toy: virtual reality. VR had basically been rubbish since Ivan Sutherland’s Sword of Damocles — the first head-mounted display (HMD) system that was so bulky it had to be suspended from the roof — right up to 2010, when a kid named Palmer Luckey turned a smartphone screen into a HMD with a wide 90° view. Then the big money came out to play. Facebook bought Luckey’s company, Oculus VR, for $2b in 2014.

Then Sony announced Project Morpheus [check out our Issue 114 feature], Samsung showed off its GearVR, HTC and gaming company Valve collaborated on HTC’s Vive, and Google made a cardboard cutout for your phone.

There are many more corporations with a vested interest this time around. Beyond the major players, there are loads of ancillary companies putting a stake in the VR biz. At the low end of VR capture, companies like Nikon and Ricoh have small dual-lens VR cameras. More productionready rigs include GoPro-style gizmos like Google’s Jump, Samsung’s Project Beyond, Nokia’s Ozo, Jaunt VR Studio’s own Neo rig, NextVR’s gargantuan RED camera rig, and VR content leader, Chris Milk’s company Vrse has its own system with integrated sound capture. On the more esoteric side — if there is such a thing in VR — non-traditional lightfield camera manufacturer Lytro has announced Immerge, and who knows what Magic Leap has up its sleeve. The company has raised almost a billion dollars with barely anything to show for it.

All of that is to say, other than AT 26

W

Front X

Right Y

X Back

Y Left Z

ERA OF AMBISONICS Ambisonics was first devised by Michael Gerzon and Peter Craven in the 1970s. It described a means of capturing sound for its native B-format, which is made up of four component channels that exhibit the responses of: W (a pressure signal corresponding to an omni pattern), X (front-to-back figure eight), Y (leftto-right figure eight) and Z (up-to-down figure eight).

With a decoder, you can create almost any polar pattern out of the B-format information, and steer it to face any direction, including dreamt up patterns like a surround sound array with three forward-facing hypercardioid and two rear-facing cardioid patterns that matches up with a typical 5.1 system. It even sums down to mono perfectly.

The most common method for acquiring this signal is via a tetrahedral microphone arrangement of four cardioid or sub-cardioid capsules.

Its native four-capsule output is called A-format, which can be converted into B-format. It’s called the Soundfield microphone, and was also developed and patented

by Gerzon and Craven. Initially it was exclusively manufactured by Calrec, before being split off into its own division, SoundField Limited.

While Ambisonics was revolutionary, it has never been a commercial success.

However, the ability to use B-format to derive binaural localisation information for spherical VR means its time is now.

(left) Sennheiser's fork-tongued, MKE440 dual shotgun mic for mounting on your DSLR is another Sennheiser audio-for-video innovation. (right) Sennheiser's Action Mic for GoPro cameras will go where normal mics shouldn't, even underwater.

Vrse’s rig, none feature more serious audio than a mono mic onboard.

While a lot of attention has been paid to acquiring VR vision, not many companies are developing tools for audio, which is where Sennheiser hopes to make a dent with its Ambeo umbrella of products.

One reason for the low uptick in VR audio technology is because we’ve actually had the means of capturing full-sphere surround sound audio, Ambisonics, well before VR vision became viable.

It’s just been sitting on the shelf waiting for its release format to arrive [see Ambisonics box].

VR REVOLVES AROUND AGAIN

Sennheiser isn’t redefining the game here, its VR Mic features the same tetrahedral microphone design originally devised by Ambisonics’ inventors Michael Gerzon and Peter Craven. Manufacturing of tetrahedral mics has been the domain of specialist manufacturer SoundField, and smallbatch enthusiasts like Coresound Tetramic and the Zoom H2n-attached Brahma. However, now that the patents have expired and the VR revolution is well underway, it’s perfect timing for the bigger players to start putting their weight behind the Ambisonics format.

“A lot of people in audio will know Ambisonics, but it was never mainstream,” said Beesley.

“Ambisonics was potentially ahead of its time because there was no real market requirement for that spatialisation of audio. With immersive technologies like Atmos and Auro, and particularly VR, Ambisonics is coming into its own.”

If VR is to avoid the same fate as 3D, content creation needs to fulfil two criteria — quality and quantity. “If the quality is poor, people won’t adopt it,” Beesley pointed out. “It’s why we feel the audio part is really important, otherwise it’s not a realistic experience.”

VR is at an odd stage. While a first-class VR experience requires proper VR goggles and a highend PC, the actual entry point into VR via Google’s Cardboard viewer for smartphones is incredibly low. You can even watch a non-stereoscopic version on your phone that still lets you spherically pan around the image while also getting the full binaural audio effect through your headphones.

The distribution infrastructure for VR is already in place too; Facebook has implemented 360 video on its timeline and YouTube lets anyone deliver content to the mass market.

Still, VR ain’t that easy to produce, so more manufacturers taking the punt developing for the format will only make it easier and more affordable. Sennheiser’s first VR Mic — which will probably have a different name when it’s released in September — will cost around €1700 for the mic and software. It’s the sort of guide price that won’t bother a moviemaker with 16 GoPros, and will sit somewhere between the Tetramic and a Soundfield microphone. “We’re trying to bring this into the prosumer market and then look at taking it into the consumer market as well,” said Beesley. If VR takes off, the entry point for soundfield microphones will definitely fall.

For now, the Sennheiser VR Mic outputs four channels of analogue audio, which is decoded in the provided software. It has a svelte shockmount that allows it to be placed in the typical VR deadspot underneath a VR rig. Besides cheaper versions in the future, Sennheiser may also work on digital versions too. “This is Mark I. We’ll see where it goes,” said Beesley. “There are conversations being had about how to develop the technology in the future. Our main goal is to make spatialised audio as easy as possible for people.

Because as soon as it gets complicated, people shy away from it.”

SPACE FOR PLUG-INS

Besides a big manufacturer like Sennheiser getting onboard the VR wagon, another sign that VR audio is being taken seriously is Facebook’s latest acquisition of Two Big Ears. Before the buyout, the company had been working on its Spatial Workstation software — a set of 3D audio spatialisation tools including AAX and VST Spatialiser plug-ins, VR video player, an encoder app, and template projects for compatible eight-channel bus DAWs (Pro Tools, Reaper and Nuendo).

The setup allows users to combine, mix, threedimensionally pan and rotate independent tracks of various B-format audio conventions, spot mics, score, voiceovers, and any other content. It also includes real-time room modelling and everything you need to encode and deliver the multiple output formats you need for interactive 3D, and stereo and binaural previews.

Facebook has since released the tools free of charge (facebook360.fb.com/spatial-workstation), along with its guide to creating VR content.

Sennheiser has also been dabbling with its own

Ambisonics was potentially ahead of its time. With immersive technologies like Atmos and Auro, and particularly VR, Ambisonics is coming into its own

spatialisation plug-ins under the Ambeo umbrella.

The Venue Modelling VST plug-in will be available mid-year, and is initially designed to virtually place DJs and producers in a handful of specific clubs around the world allowing them to pre-experience their sets from the audience’s vantage point. It seems like a bit of a gimmick, especially when you consider the limited audience for a plug-in like it.

However, it makes sense as part of Sennheiser’s overall 3D push. If VR takes off, being able to model performance venues could create an entirely new way of experiencing concerts remotely. There’s also no limit to how this technology could be applied in other areas of VR content creation.

GOING PRO

Not content to just go long on the VR business, Sennheiser has been looking further afield and developed some novel cameratop mics for the current crop of filmmakers. There’s the forktongued MKE440 stereo shotgun, which isn’t a configuration you typically come across. Its roughly 45-degree splay gives you a slightly wider spread than a typical mono shotgun, but without losing definition like a standard XY cardioid setup.

However, the most exciting piece of gear Sennheiser will shortly release is also its smallest.

After evaluating the market, the mic manufacturer saw a big gap in the action mic category. There was just nothing there that could stand up to the abuse those cameras are subjected to.

“We were generally looking at different markets. The main one was people producing AT 27

footage for Periscope and YouTube, and it became obvious to look at action cams,” said Beesley. “We decided to go for the biggest one, GoPro. One of the biggest GoPro end user complaints was the sonic quality. Because it’s quite an investment, the first thing people do when they buy a GoPro is put it into the protective waterproof housing. As soon as you do that you lose the ability to record decent quality audio.

“An awful lot of footage on YouTube is either dubbed over or set to music. We started the conversation with GoPro right at the point they were beginning to formulate their Developers Program to look at different accessories. They invited us to join their program on the audio side.

They had a big launch in April with 100 developers and we’re the only audio developer in the program.”

The Action Mic is Sennheiser’s first foray into the action cam-specific market. The requirements were obvious, yet daunting. It had to be water resistant enough to dive with, wind resistant enough to handle the speed of a motorbike, and rugged enough for extreme environments.

“The mic itself is an omnidirectional electret capsule that’s been developed specifically for that microphone,” said Beesley. “The windshield is a new technology and unlike normal windshields.

The dead ferret type of affair is very effective, but when they get wet the hairs lay flat and you lose effectiveness. This is a new type with stiffer filaments that don’t lie flat when wet.

“The major development in small electret microphones has been to make them sweat resistant for theatre and be able to clean makeup off the microphone without damaging the element. It was a progression from the technology we already have for sweatproofing our microphones.

“The membrane used on the surface of the element is acoustically transparent, but it also has a surface tension that doesn’t break the surface tension of the moisture drop. Therefore it’s impermeable to liquid; you can literally take it down underwater. Whenever you say a mic is waterproof, you have to be very careful; water resistance is probably a better term. Because of the nature of the membrane, if you dive deep with the microphone, the pressure can force water through the membrane. The microphone we’ve produced is not for deep sea diving, but certainly anyone doing water activities above a 10m depth should be absolutely fine.”

One of the benefits of being part of the Developer Program is access to GoPro’s preamp and circuit specifications without having to reverse engineer them. It allowed Sennheiser to craft the best possible response that matched the GoPro field of view with the most usable sensitivity. “We really thought about what people want to capture,” said Beesley. “Firstly, they want the environmental noise, the sound of the skis or wakeboard, but they also need the ability to hear what they’re saying. A lot of people put action sports on YouTube to show

Sennheiser's VR Mic looks pretty normal with its head basket on.

people what they’re doing and comment at the same time. With this, rather than having to mask it with music, they can put it straight onto YouTube with their own commentary and environmental sound, so people can get a more realistic experience from the video.”

In the same way that a stereo shotgun brings a new perspective to cameratop audio, Sennheiser will release other action mic variations in the future. “This is our first microphone,” said Beesley.

“But there will be others in the range for GoPros.”

Hopefully having Sennheiser push the boundaries of where a mic can go will push other manufacturers to do the same.

CLAIR BACKS JPJ AUDIO At the end of last year, Clair Global invested in a controlling share of JPJ Audio. We sat down to talk about it with JPJ Audio Director/Shareholder, Bruce Johnston, who was surprised the news had stayed under wraps for so long. “As nothing was really changing other than a new partner, it was business as usual,” said Bruce. After he sent an official statement about it to all his staff and main contractors, he told AT, “All the feedback he’s got has been largely positive.”

According to Bruce, the decision wasn’t a light one: “It became obvious towards the middle of last year that Eric [Robinson, CEO of JPJ Audio and major shareholder Jands Production Services] was too sick to come back. He hadn’t been in the office for 12 months.” Eric sadly passed away November 11, 2015 from complications with cancer, but before he did, he wanted to make sure JPJ was in good hands.

Bruce: “Eric approached Troy Clair, they’ve been mates for 35 years, and asked if Troy wanted to buy JPS’ part of the company and keep the JPJ legacy going, after all it’s been going for 40 years.

Troy thought it was a good idea, and we worked forward on that. The other options weren’t in the best interest for staff or the industry. Troy said JPJ Audio was a great company and why would he AT 28

want to change that. He pointed out that being able to use each other’s inventories in the off-season would be advantageous, rather than spending more money on gear.”

Other than a global warehouse of inventory to call on, another side effect of Clair’s involvement in JPJ is the ability to tap into Clair’s IP and the opportunities it opens up for staff of both companies to work overseas. “Hopefully we can get them on tours around the world,” said Bruce.

“It’s been easy for us to bring people here, but very difficult for us to send them to America.” Both those things will help JPJ Audio compete in the ever changing-world of audio, with the arrival of

more international players. “There’s always going to be competition, and it only makes us better,” said Bruce.

The day to day running of JPJ Audio remains the same under CEO Jim Straw, and Bruce is still a Director/Shareholder, alongside Clair Global.

Meanwhile, the structure of the company has become more unified with the previously separate entities of theatre and recording trucks now under the JPJ Audio banner. Bruce still has Johnston Audio as a separate install entity. “I’m in the process of moving to just supplying d&b and L’Acoustics in my installs now,” he said. “There’s so much competition, I’m just going to supply the best.”

AT 29

FEATURE

ORCHESTRATING A BEATLES SHOW The Beatle Boys upped the ante for tribute bands everywhere by adding orchestral backing to their live show.

Story: Preshan John

If you’re in a band, I guess it’s the ultimate compliment knowing there are people out there who play covers of your songs. It’s also pretty flattering to think that’s all some groups do — play your music. But does it get creepy when they try to look the same as you? Dress the same? Mimic your facial hair?

Sure, tribute bands can be cringeworthy, but there are some that do more than just replicate the look of their icons. The Beatle Boys from Sydney are one such group; they’re a four-man outfit that sounds the part as much as they look it. While I’m sure it would positively freak John Lennon out meeting his impersonator (who looks uncannily similar), it’s hard not to feel like these guys are the AT 30

real deal when you hear them live.

These days The Beatle Boys are going above and beyond, augmenting the band’s original arrangements with orchestral backing for their The Beatles Orchestrated stage show.

The original arrangements were written by renowned conductor George Ellis, and during the Australian tour Ellis hand picked each of the city’s 30-plus orchestra members himself. AT attended the Melbourne show in Hamer Hall; it was a sell-out gig with an audience of over 2000 in a venue that puts a stamp of authenticity on anyone that plays there. Dean Lovell was at the helm of a Yamaha CL5 console, mixing the show over the Arts Centre’s Meyer Mylo line array system.

FILL UP THE FRONT A series of Meyer M1D line array elements line the stage as front fills for the orchestral pit which was an overflow seating area for The Beatles Orchestrated. You’ve got to show off your orchestra if you’ve got one.

PERSONAL MONITORS Yamaha MSP5 speakers are dotted about the stage providing personal monitoring for the pianist, key orchestra members, and conductor. The band uses IEMs. Lovell takes care of both FOH and monitor mixing, and with 13 individual sends, he’s got his work cut out for him.

56 IN, 48 OUT Familiarity is the main reason Lovell sticks to Yamaha consoles. The CL5 manages 56 inputs that come in via a Danteenabled stage box. Some orchestral inputs are submixed down to stereo before the whole lot goes into a 48-channel Tascam X-48 standalone multi-track recorder.

ONE-MIC JOBS The classic AKG C414 ULS is perched over the timpani. Lovell mikes all orchestral instruments in mono only. A Neumann KM100 is gaffed to the bridge of the closed-lid Yamaha grand piano and the harp is also miked with a single KM100

Neumann KSM105s are the vocal mics of choice for the band, and each is named according to its Beatles member. This one on stage right is dubbed ‘the McCartney mic’.

AT 31

ZERO AMP STAGE The electric guitars feed directly into the console — one through a Line 6 POD HD500X processor, and the other through a pedal board equipped with Tech21 amp simulator boxes. The bass guitars are also DI’d direct. Lovell says having amps on stage would make life difficult for the orchestra and conductor.

Lovell is a big fan of the Beyerdynamic TG Drum Set Pro microphone pack. The drum kit calls for just five mics: the TG70d on kick, TGI53c small diaphragm condenser on hats, and the TG58c clip-on condensers on snare and toms.

Two ground stacks of five Meyer M’elodie array elements flank the stage as side fills, along with a Meyer UPA-2P point source loudspeaker and companion sub.

AT 32

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RF Transmission Systems The most common of the wireless transmission technologies, RF transmission provides the most powerful system with the largest range of coverage. The nature of RF signals allows users to leave the room in which the transmitter is installed without loss of coverage, and the system is also well suited to outdoor events. The receivers can be equipped with tele-loops which provide for the inductive transmission of signals to the hearing aid.

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AT 33

TUTORIAL

u o y o d w o H ? p u e r u s a e m III PART

The Difficult One Frequency at a Time There’s more to phase than just flipping it. Here’s how to read phase when it starts getting shifty.

Review: Ewan McDonald

When I first started using a measurement system, I would hide the phase response because I had no idea what it meant. It wasn’t until someone actually spent the time to show me how the phase trace worked that I realised how useful a tool it is. These days I wouldn’t consider measuring a sound system without it.

We understand the concept of frequency response long before we become sound engineers.

Every music listener can hear the effect of turning the treble or bass dials. On the other hand, it’s virtually impossible to determine the phase response of a signal just by listening to it.

Don’t worry, in the realm of system design and alignment, the audibility of the phase response of a single source is the least of your worries. What is going to be really useful is learning how phase plays a role when two or more sound sources interact.

AT 34

PAST THE DIFFICULT PHASE

A sound engineer’s first encounter with phase is generally in the form of a polarity inversion, which flips the phase 180° at all frequencies. Things usually go astray when we start talking about phase shift.

In reality, phase isn’t actually hard to understand.

You were probably well underway to understanding it in you first audio lesson, you just may not have learnt how to apply it in the real world.

Let’s start with the sine wave. It has a frequency, amplitude and phase component, and in our case, represents the compression and rarefaction of air particles created by a sound wave. Easy, right? Well the problem is, that’s where most explanations of phase stop, so we struggle to translate theory into the real world where signals are anything but pure sine waves.

The connection between sine waves and the real world lies with a long deceased Frenchman by the name of Jean-Baptiste Joseph Fourier. Sound familiar? Probably because Fourier is the middle ‘F’ in FFT (Fast Fourier Transform), the type of algorithm used in most measurement systems today. What Mr Fourier discovered way back in the early 1800s is that a complex waveform can be broken down and represented by a series of simple sine and cosine waves. This goes for anything we hear, it doesn’t matter if it’s a cello or a heavy metal band, if we look at a single moment in time, we can break down that signal into individual frequencies and calculate the amplitude and phase at each of those frequencies. Even distortion components like square waves can be broken down into an infinite sum of sine waves.

5

3.75

When I first started using a measurement system, I would hide the phase response…

These days I wouldn’t consider measuring a sound system without it

2.5

Wave 1

Ampitude

1.25

Wave 2 Wave 3

0

Wave 4 Wave 5

-1.25

Result

-2.5

-3.75

-5

ONE FREQUENCY AT A TIME

If you’re ever overwhelmed by phase, remember that even measurement systems and loudspeaker simulation programs do their work one frequency at a time. When you look at your transfer function, the measurement system has used this Fourier Transform to break down the signal into its individual frequency components and calculated the amplitude and phase shift for each of those frequencies. It then plots this information on our transfer function graph and essentially ‘joins the dots’.

When we look at loudspeaker simulation software, for example, the software has not only done all the calculations one frequency at a time, but it has made each of these individual calculations for every sound source, one position at a time. It can then sum the results at each individual frequency to display them over a wide bandwidth (e.g. Full range A-weighted or 2-8kHz).

The easiest way to wrap your head around complex phase interactions is to think like your computer — one frequency at a time, one position at a time. Once you understand this concept, everything gets much simpler to grasp.

Before we get out our measurement kits, let’s start by looking at the simplest of phase shifts — the time delay. If we shift a wave in time, we will shift its phase relative to the original wave.

500 Hz 1

0.75

0.5

0.25

Ampitude

FUN FACT: Why does the Fourier Transform use sine and cosine waves? Well, a cosine wave actually looks identical to a sine wave of the same frequency. However, each point of the cosine wave occurs exactly 1/4 of a cycle earlier than the corresponding point on the sine wave. By varying just the amplitude of the sine and cosine waves and summing them together, we can create a new wave with any phase shift and amplitude at that frequency.

Original Wave

Wave shifted by 1msec Polarity Inversion

0

-0.25

-0.5

-0.75

-1 0

1

2

3

4

Time (msec)

SHIFT BY EXAMPLE If we add 1ms to a 500Hz sine wave, let’s calculate how much phase shift is applied relative to the original sine wave. The time period of 500Hz is 2ms (i.e. 1/500 of a second), so our 1ms delay creates a phase shift of 180° relative to the original wave. A delay of 0.5ms would shift it by 90° and a delay of 1.5ms would shift it by 270°. A 2ms delay would obviously shift our 500Hz wave by a full cycle and this wave would look identical to the original. This part should be relatively easy to understand. If you can’t grasp this concept yet, go back over it until you do. You’ll need to understand this before moving on.

Now let’s use that same 1ms delay, but apply it to a full range signal, like pink noise. As every

frequency has a different time period, every frequency will have a different phase shift, so we have to break it down frequency by frequency.

IMPORTANT TO NOTE: UNLIKE OUR SINE WAVE, THERE IS NO SINGLE NUMBER FOR A PHASE SHIFT OF A FULL RANGE SIGNAL. WE HAVE TO LOOK AT THE PHASE SHIFT FOR EACH INDIVIDUAL FREQUENCY.

We’ve just worked out that with a 1ms delay our signal will be shifted 180° at 500Hz, but what about 1kHz? Well, the time period of 1kHz is 1ms, so we will create a phase shift of 360°, or a full cycle, at 1kHz. If we look at 250Hz, which has a time period of 4ms, we’ll be shifted by 90° relative to the original wave.

AT 35

1ms After Reference

Phase 150 120 90 60

Starts at 0°

Back to 0° at 1KHz

30 0 -30 -60 -90 -180° at 500Hz

-120 -150 31.5

63

125

What does this mean to us in the real world?

Well, for a single source and a time delay, absolutely nothing. We’re only talking about ‘relative’ phase here. No amount of time delay to a single source will change the way that source sounds, only when it arrives. Where this information becomes useful is when we need to add it to another sound source.

Knowing the relative phase shift between two sources is crucial in determining how they interact.

PHASE, IT’S ALL RELATIVE

It’s almost time to get out our measurement systems, but first we must understand how the phase trace is calculated. When phase is displayed on a transfer function, we’re looking at the phase response of the measured signal shown relative to the reference signal. Your measurement system should have the ability to calculate the delay of a measured signal relative to a reference and then insert that delay into the measurement system. If you don’t do this your phase trace probably looks like a jumbled mess.

When you press this delay finder button, the measurement system takes an impulse response of the system. It finds the initial time arrival using the initial peak of the impulse response, and adds that delay to the reference signal so now the measured signal and reference signal arrive at the measurement system at precisely the same time. It then compares the two signals and gives us the magnitude (frequency) response and phase response of the measured signal, relative to the reference signal.

When the measurement system displays phase, it does it in a -180° to +180° window. The reason for this -180° to +180° window is because measurement systems break a signal down into individual sine/cosine waves before displaying it, and since a sine or cosine wave with a 360°, 720°, 1080°, etc phase shift looks identical to the original wave, the measurement system cannot determine if the wave has gone past this 360° window. This is what we call ‘wrapped’ phase response.

AT 36

250

500

WRAPPED VS UNWRAPPED PHASE Most measurement systems have the option to ‘unwrap’ the phase response. This extends the phase graph beyond ±180° and allows you to view the phase trace as a single continuous line. Using this for alignment is a mistake, not only because the graph becomes compressed making it harder to read phase differences accurately, but it is prone to errors due to the way it is calculated. Use with caution!

When we look at an unwrapped phase response, the measurement system actually has to work out when the phase is jumping from +180° to -180° (or vice versa) and then splice the data together to make a single line. Any errors in the measurement can be compounded when the phase is unwrapped.

This works fine in electronic measurements, but errors are introduced as soon as you add a loudspeaker and environment. Some measurements also have a group delay option, but this has to be calculated from the unwrapped phase, meaning those same errors also occur in the group delay.

I’ve seen many people (myself included) look for fancier ways to use their measurement systems, but unless you’re working in very controlled environments, know exactly what you’re looking for, and have a specific use for group delay or unwrapped phase, stick to the regular old boring wrapped phase.

GETTING SET UP

We’re going to use our example of a 1ms delay and look what that does to the phase response of our signal, but first we need to get set up. The best way to do your initial phase measurements is electronically (i.e. not using loudspeakers), since the loudspeakers and environment will cloud your results. You’ll need to take the noise

1k

2k

4k

signal from your measurement system, split it into two channels, process them independently, then combine those two signals back into one, and feed that into your measurement system.

If you work in live sound, grab a Y-cable, a speaker processor with the ability to adjust delay on each band and a mixing console to mix the two channels back into one. If you work in installations, a configurable audio DSP that can mute, delay and mix internally will work a treat. Since we’re making electronic measurements, you can turn off your trace smoothing and averaging.

First, we’re going to look at a single full range signal, so mute one of the channels. Run your measurement noise signal through your processor/ DSP setup and back into the measurement system.

There will be some form of small delay through the system, so run your delay finder and insert the delay. If your gain is adjusted right — and your system is free of filters (intentional or otherwise) — you should see two straight lines; one in the frequency display at 0dB and another in the phase display at 0°.

If your phase trace is sitting on the 180° line then you have a polarity inversion (relative to your reference) somewhere in your signal chain. Make sure your frequency response is sitting at 0dB and the phase response graph is centred so that 0° is in the middle with -180° at the bottom and +180° at the top. This will make it easier to read relative differences later on. Don’t be deterred if your phase trace isn’t perfectly flat at the extreme low and high frequencies, some electronics will have a slight phase shift at either end of their frequency spectrum. As long as it’s flat for most of the useable frequency range it will be fine.

Now insert a 1ms delay into your crossover or DSP. YOU MUST LEAVE THE INTERNAL MEASUREMENT SYSTEM DELAY THE SAME, SO IF YOU HAVE AN AUTO TRACKING DELAY FUNCTION THAT RECALCULATES THE REFERENCE DELAY EVERY TIME THE SIGNAL MOVES, YOU’LL NEED TO ENSURE THIS IS TURNED OFF. The frequency

response should stay the same but the phase

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response should have shifted to a line that slopes downwards from low to high frequencies. When it reaches -180° it disappears and pops out at the same point at the top of the screen (+180°) and continues like this getting steeper and steeper as you go higher in frequency. This is the ‘phase wrap’ occurring. Older Smaart systems continue the line directly between the top and bottom point (which creates a kind of ‘shark fin’ or sawtooth looking trace) but these vertical lines actually serve no purpose, so I prefer it without them.

Going back to our example, a 1ms delay should give us a 180° phase shift at 500Hz, so let’s compare

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Signal After Reference

that to our phase trace. We can see that the line starts around the 0° mark in the lowest frequencies and then slopes downwards until it hits the -180° point at precisely 500Hz. If we continue along that line, which should now start again from the top of the screen, it should continue downwards crossing the 0 deg line at precisely 1kHz. This is because the time period of 1kHz is exactly 1ms, so a 1ms late arrival is phase shifted by 360° at 1kHz. Since this is the first time it has ‘wrapped’ around and passed the 0° mark again, we know that a 360° phase shift has occurred. The second time it crosses the 0 deg line is at 2kHz meaning that 2kHz has a phase shift

of 720° relative to the reference signal. Since 2kHz has a time period of 0.5ms, this makes sense.

Now capture that trace and increase the time delay in your processor to 2ms. What we can see is that the phase trace is now much steeper and wrapping from -180° to +180° more often. Let’s again follow the trace from the lowest frequencies to the point it first hits the -180° mark. This should be at 250Hz. We should know from our previous example that 250Hz has a time period of 4ms, so a 2ms delay would put the phase shift at 180°.

Understanding that the phase trace becomes steeper when the measured signal moves further away from the reference signal will be important down the track, so try and ingrain this in your memory.

Now what happens when the measurement trace is earlier than the reference signal? Since there’s no way to add negative delay to your processor/DSP, we’ll need to add delay to the internal reference delay in our measurement system. Since we should now have 2ms in our processor/DSP, take note of the reference delay in your measurement system, and now add 2ms to that number (NB: don’t simply enter 2ms, you will need to add 2ms to the time already stored as your reference delay). This should bring your measured signal and reference signal back into alignment and the phase trace should return to the 0° line.

Take note of this. The sound source has not moved at all, just the internal reference in the measurement system.

Now add a further 1ms to the internal reference delay in your measurement system. You should see a line that is a mirror image of the previous trace with 1ms on the measured signal that moves upwards from the low frequencies and hits its +180°mark exactly at 500Hz. This upward slope means your measured system is before (leading) the reference signal. If it’s sloping downwards it is after (lagging) your reference signal.

The way I used to remember this when I first started is to imagine a mountain with a flag on top, and a climber trying to reach that flag. The flag is the internal measurement reference, the mountain climber trying to reach that flag is your measured signal, and the slope is the angle of your phase trace. Before the climber has reached that flag — i.e. the measured signal is before the reference signal — they’re on the ‘up’ slope. Once they’ve passed that flag — the measured signal is after the reference signal — they are now descending the slope. When the measurement signal and reference signal are arriving at the same time, the line is flat, like the climber reaching a plateau at the summit.

HOW TO USE IT

Nice information, right. But how do you use it?

This phase trace is going to help you determine how two sound sources are going to interact when they’re added together. The important thing to understand here is that what you’re doing is comparing two individual phase traces to the combined frequency response trace of both signals.

This is a fundamental concept in sound source interactions — i.e. phase responses of those AT 37

If you’re ever overwhelmed by phase, remember that even measurement systems and loudspeaker simulation programs do their work one frequency at a time

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individual sources will help determine the outcome of the frequency response of the two signals combined. We should already know if we add two sine waves that are out of phase (180°, 540°, 900°, etc) we should get a complete cancellation. If they’re in phase (0°, 360°, 720°, etc) we should see an addition.

Let’s look back at the 1ms example. If we just look at the 1ms phase trace and think what would happen when one source arrived at a point exactly 1ms later than the other source, every time that trace hits the 180° mark, we should see a cancellation, and every time it hits the 0°mark we should have an addition. This is what we call the comb filter, and if you’ve read the first tutorial in this series, you should be familiar with this already. To see this occurring on the screen, start by resetting your processor delay, then set your measurement delay so you’re back to 0dB in the frequency graph and 0° in the phase graph. Now add the second channel in your processor/DSP. Don’t touch the delay yet. Your measurement trace should now lift by 6dB in the frequency graph but remain at 0° in the phase trace. If it doesn’t, the delay (or polarity) between your two channels is not identical. Now add 1ms to only one of those channels on your measurement system and you should see a very nice comb filter. Don’t bother paying attention to the phase trace once both channels are mixed together.

As an experiment, leave your internal reference delay where it is, and give one signal 3ms delay and the other 7ms delay. Measure these individually without changing the internal measurement delay and capture the traces. Now you’re comparing to more than the 0° line, so you’ll need to compare the two traces to see how the two sources will interact. Overlay the two of them and look at where the phase traces cross over each other. At all these points of intersection the two signals are in phase and summation will occur. Wherever the traces are 180° apart (e.g. -50° vs 130°) then cancellation will occur.

Now let’s go to our basic polarity inversion.

Can you guess what happens to the phase trace of a delayed signal if the polarity of your signal is flipped? Any time we have a polarity inversion between two sources, the phase trace will be the same shape but 180° out at all frequencies. One AT 38

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great use of the phase trace is spotting polarity inversions in signal chains, especially when measuring multi-way loudspeakers.

DELAYED GRATIFICATION

Now let the experimenting begin. Start playing around with both the signal delay and the reference delay. Get a feel for how the trace reacts. If you’ve got someone with you — put your reference delay to an arbitrary amount and get your partner to add delay to your measured signal (and not tell you the amount) both above and below the original amount. If you’re okay with maths, you can actually calculate what that delay was just by looking at the phase trace. If not, just try and get a sense of when your measured signal is getting close (i.e. within a couple of milliseconds) as opposed to far off.

Start to also get a feel for how the steepness of the curve changes as the delay between the measured signal and the reference signal increases. Now add different delays to both of your input signals, measure them individually and before combing them, take a guess at how the frequency response should look when you add them back together. These types of experiments will not only get you ready to take your phase trace out into the real world but help

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give you a fundamental understanding of phase interactions between sources.

In the next article, we’re going to continue our journey into the world of phase and look at phase shifts that occur from changes in frequency response and how to use the phase trace to start to combine sources with different frequency ranges like a full-range speaker and a subwoofer. We’ll also start taking a look at real world measurements and ultimately learn how to use the phase trace to make your systems sound better!

BIO Ewan McDonald has measured just about every manner of PA system all over the globe from boardrooms, dingy pubs and old churches right up to opera houses, 10,000+ seat mega churches, and arena and stadium rock PAs for some of the world’s biggest artists.

After spending many years as a senior system engineer at Norwest Productions and an Applications Engineer at Adamson Systems Engineering in Canada, Ewan has since settled back in Australia as a System Sales Specialist at Technical Audio Group (TAG) in Sydney.

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youtube.com/yamahaaustralia AT 39

REGULARS

PC Audio Struggling with overheads?

Follow this guide and they could soon be a thing of the past.

Column: Martin Walker

A reader recently emailed me to ask which free utilities I recommended to manage or control Windows Services and Processes, since the CPU overheads on his audio PC were hovering around 52%, even before he launched any applications. I replied fairly quickly to the effect that I wouldn't advise him to attempt to disable any Windows Services, since in my experience this rarely results in any measurable improvements in performance, yet can seriously impinge on the reliability of your machine.

For example, I once spent an entire afternoon disabling a total of some 46 Windows Services on my resolutely audio-only PC partition, carefully following the advice of a well-known commercial builder of Audio PCs. After a reboot I re-measured its performance, only to find that while the system RAM consumption had dropped by a miniscule 8MB, its CPU overheads hadn't changed one iota, and nor could I measure any improvement in performance on any audio benchmark tests.

Moreover, during the following week I experienced several infuriating crashes, and even found a repeatable sequence of actions that would crash Cubase every time, requiring a reboot. Fortunately I'd created a hard drive image immediately before these Service tweaks. I restored this image and all my crashes ceased immediately.

OVERHEAD CHECKLIST

Given that disabling Windows Services can result in instability or even a complete refusal to boot up afterwards, what else should this reader have been exploring to resolve his 52% overheads?

Well, the most important thing to check is that your computer's Power Plan is set to 'High Performance'. The wrong setting here can cripple the processing performance of many modern PCs, because of over-clever power schemes that throttle your processor to a slower clock speed to keep it cool and, in the case of laptops, prolong battery life. In theory, such throttling schemes should let your CPU clock speed ramp up smoothly on demand, but in practice there's a short time lag before this happens, sufficient to result in audio interruptions and, therefore, clicks and pops. The AT 40

only safe way to prevent this happening is to make sure your processor always runs at its top speed, so that your PC doesn't unexpectedly conk out during song playback.

I still disable all System Sounds because you can never be too sure what will happen to your DAW if Windows suddenly decides to (for instance) emit a ping to inform you of an incoming email. Many system WAV files are still recorded at a sample rate of 22kHz, and depending on your audio interface, your currently running 44.1kHz audio project may suddenly jump an octave higher and run at double speed, and your soft synth playback become extremely 'juddery'. Other musicians have reported songs dropping to half speed, playback stopping altogether, or suffering ongoing timing problems, requiring a frustrating reboot.

I also still change Processor Scheduling to 'Background services', as originally recommended

by Steinberg for anyone using its ASIO drivers (and nowadays that includes just about every PC musician), since ASIO drivers run as a background service in Windows. Although this particular tweak does cause some debate among experts (some claim it makes no difference nowadays, while others still maintain that you may be able to run your audio interface at a significantly lower latency afterwards), I've never experienced any detrimental effects, so continue to use it with care. However, from Windows 7 onwards, most of the other published tweaks recommended for musicians provide rapidly diminishing improvements. By all means explore some of them if you're determined to squeeze the very last drop of performance from your PC, but often the improvements (if any) become too small to be reliably measured.

YOUR NEXT TASK

If your DAW is running a lightweight project that nevertheless results in high processor overheads, it may simply be that you've chosen a particularly low buffer size for your audio interface. Once your audio latency drops below about 12ms (512 samples at 44.1k) CPU overheads will start to rise, and may do so radically below 6ms. To rule out your interface from overhead concerns, just switch

its buffer size temporarily to 512 samples and see if your problem goes away. If your PC still displays high overheads even when it appears to be running no major applications, the next port of call is to open the Windows Task Manager and click on its Processes tab, to explore just what is tying up your processor — you'll see quite easily from the CPU column which processes (threads of code running in the background) are consuming the most cycles.

Once you've tracked down the culprit(s), just rightclick on the process and select the 'End Process' option to check that this cures the problem.

‘But how do you know what application launches a particular process?’ I hear you cry. Well, an internet search engine will help. Simply enter the full name of the process and you'll soon find out not only what it does and the likely application that installed it, but also how to remove it permanently if it's causing problems. This is an inexact science, so go slowly but surely, dealing with one high CPU process at a time until your computer's CPU overheads drop to less than a few per cent when idling.

The most frustrating processes tend to be those that get automatically run each time you boot up your PC, such that they keep popping up even after you've disabled them. The quickest way to see a list of what Startup Items are currently installed on your PC is to run the msconfig.exe utility from the Windows taskbar and click on its Startup tab.

Another option, and the one I prefer, is to use the Startup option in the Tools section of CCleaner (a free utility download from piriform.com), since this displays the info far more clearly. A handy web site that documents a huge number of startup tasks and what they do can be found online at www.pacs-portal.co.uk/startup_search.php, but most should be fairly obvious from their entries in CCleaner's 'Program' and 'Publisher' columns (or the Msconfig 'Startup Item' and 'Manufacturer' columns). You can temporarily Disable any startup task so that it won't get re-launched on the next boot, and if this solves your problem then the final step is to permanently delete it. Good luck, and may your overheads plummet!

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AT 41

QUICK MIX

The

with

James ‘Jimbo’ Barton

When did your audio engineering career begin?

I have been engineering and producing for 30 plus years! Bill Armstrong (AAV/Armstrong Studios Melbourne) hired myself, Quintin Rosario and Demo Karabatakis to work in his ethnic radio stations 3EA and 2EA. It was a great learning process but I found myself constantly sneaking around to the music studios every spare moment I had. I eventually begged Roger Savage (Bill’s partner and my mentor) to let me transfer to the music studios. He told me it would be hard work, long hours and no social life, but he did say yes. It quickly became everything he said it would be, but I was surrounded by amazing teachers; Roger, Ernie Rose, Ian Mackenzie, Ross Cockle and a host of incredible musicians, writers, producers and bands. I ended up doing midnight to dawn sessions with Mushroom Records bands, and that was the start.

In 1982, after working with Australian artists such as the Little River Band, Skyhooks, Jo Jo Zep & the Falcons, The Sports, Split Enz, The Models, Paul Kelly & The Dots, Mondo Rock, Cold Chisel, Australian Crawl, Jimmy & The Boys, I moved to London and worked at Sarm Studios with Trevor Horn and Peter Collins before relocating to the USA in 1991.

Where are you currently working and what is your current method of recording/mixing?

I live and mainly work in Los Angeles, but I still travel quite a bit for projects abroad and my preferred recording platform is Pro Tools.

If I'm tracking drums I like to use East West Studios in Hollywood.

I externally compress and EQ (1176 and Neve) the signal on the way in. I've recorded many projects on their beautiful old Neve. The post process is arranged according to budget.

Out of the box, it's usually at Reel Music Studios in Marina Del Rey, which belongs to my partner and long-time friend J.J. Farris. He has a full 72-channel SSL 6000E console in his downstairs guest house, that just happens to boast a pool right through the double French doors of the control room, and a panoramic ocean view. It's a tough hang!

For mixing in the box I use a studio close to home downtown, Silverlake Pro, run by another good friend, Chad Gendason. He has an SSL Sigma rack mount. I use an SSL compressor and an NTI EQ3 over the stereo bus in either situation and mainly use SSL G EQ plug-ins.

AT 42

I was apprehensive mixing in the box at first, but I've since mixed many projects at Silverlake Pro. The incredibly talented and dear friend, Cliff Maag, who designed and built the original NTI EQ3, has now become Maag Audio, and has a wealth of other amazing toys. I'd used Cliff’s EQ for over 20 years before we finally met.

What are some of the projects you’ve worked on in the last 12 months?

The last two years or so I've moved more toward sound design for film. This year, indie films Breakfast With Curtis, City Tears, A Southern Tale and a series of trailers for the forthcoming Bourne Identity movie with Matt Damon, which I mixed at Michael McGlynn’s Vienna People studio in Annandale, Sydney. I've produced a Matchbox 20 DVD, a Queensryche album and some 30 Seconds to Mars surround mixes.

What are some of the bands you have worked with, or projects you have worked on in the past?

Credits vary a lot and too numerous to list all here. Earlier projects in London were Kate Bush, Garry Moore and Phil Lynott. Later came Freddy Mercury, Eric Clapton, Enya, Phil Collins, Rush, John Lydon (PIL), David Bowie, Thomas Dolby, Howard Jones, Voice of the Beehive, Gene Loves Jezebel.

What’s your ideal recording space and method?

I don't really have one. I walk into a room and roll with whatever equipment they have. I think it helps make every project have its own sound. Having said that, I do need a pair of Neves, the NTI equaliser and an SSL compressor.

Favourite microphone or any other pieces of kit that are indispensable to you?

I'm a Neumann dude, but I do keep a bunch of Shure SM57s and Sennheiser 421s around.

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In the last 20 years, what are three pieces of gear that have been game changers for you?

NTI products, Pro Tools, plug-ins and keyboard instruments. There is so much great stuff out there, I can't really list my faves.

How have your working methods changed over the last 20 years?

Outside of incredible equipment transitions, not much at all. When I walk into pre-production with a band, I already hear the final mix in my head.

Most memorable session or career highlight?

Ironically, some of my best memories are my early years in Oz. Mondo Rock, Split Enz/Crowded House, The Sports, Jo Jo Zep, The Models, Hunters & Collectors, etc. It was a really exciting time in Oz music, but I would have to say that Freddy Mercury made an indelible impression on me. The single most creative musician and accomplished vocalist I've ever been lucky enough to work with.

When we finished the project I was invited to Paris for the opening show of Queen's Magic tour. Freddy was an incredibly generous man and they're amazing memories.

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Any tips/words of wisdom for someone starting out?

If there were no such thing as illegal downloads, I would say learn your craft well, but at this point, save your parent’s money and stay in college! Sad but true.

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AT 43

REVIEW

DIGITAL AUDIO LABS LIVEMIX Personal Monitoring System Livemix lets performers craft their own mix in as little or as much detail as possible. From just a bit more Me to delving into the onboard EQ, dynamics and effects via the full-colour touchscreen, Livemix suits everyone.

NEED TO KNOW

Review: Mark Davie

PRICE CS-Duo - $999 Mix-16 - $1899 Mix-32 - $3625 AD-24 or Dante Card $1715 FP-2 - $125 Complete bundle (incl. 4 CS-Duos) - $7995

AT 44

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

PROS Full-colour touchscreen enhances operation Onboard ambient & intercom mic Built-in EQ, dynamics & FX processing Simple to operate & sounds great

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SUMMARY Even amongst performers armed with digital console tablet apps, the dedicated personal monitor mixer still stands tall. Livemix breathes new life into the concept with even more detailed control via its full-colour touchscreen and two separate mixes per unit. It’s got everything you need onboard to get ultra personal with your mix and interact with everyone and everything around you.

You could hear it coming, bubbling up like a slow rising tide, but there was not much I could do. I’d just handed control of an entire stage’s worth of monitoring to a haphazard bunch of musicians, and they were already fighting each other for onstage level.

This was the first time I’d pressed Digital Audio Labs’ Livemix system into service, and it had been going swimmingly until this point. Livemix is a personal monitor mixing system; think Aviomstyle personal mixers on the end of an ethernet distribution network. It had been dead easy to set up, only taking me just over an hour to integrate the system into a setup that had previously required mixing monitor sends from FOH. That included: wiring the system between the stage rack and monitors; running ethernet cables; attaching mixer units to mic and music stands; re-building a Digico session to include more aux outputs, re-routing all those aux outputs to the stage rack; pre-mixing drum sends and sending everything else to its own aux; then getting my head around the units and dialling up five individual monitor mix starting points.

Now, Livemix will more likely find a home as a companion to an in-ear system where stage sound won't be an issue. Even when pressed into this sort of mixed in-ear/stage monitor scenario it does have some built-in measures to stop this stage sound creep. Things like a cap on each unit’s master output level, which I thought I’d set to a reasonable level. Obviously my generosity exceeded common sense. In my defence, it’s hard to predict this sort of psychological push-’n’-pull gain structure until all the instruments are playing, and there wasn’t much time to get that balance just right before action time.

Despite a bit more monitor mud than I would have liked, Livemix’s integration had been a success. The musicians loved the ability to control their own levels, they found it easy to use after just a five-minute walkthrough — one singer, who I know to be a bit of a technophobe, was even schooling others on the system. At the end of the show everyone decided they needed it in their lives.

Case closed? Well it’s looking good for Livemix, but let’s get into some detail.

TABLETS: WRITING ON THE WALL?

It’s been a while since Aviom first tried to pry monitor mixes out of engineers’ hands, all the way back in 2002 when it released the A-16. Since then, digital consoles have become the norm in the live world, even in your average install situation

like schools and churches. Almost all the new breed of digital consoles also offer tablet or phone control apps, which are not only easily configured as monitor send controllers, but often offer features specific to that task like user lockouts and cutdown monitoring-specific phone apps.

It begs the question, does the dedicated personal monitoring mixer have a place onstage anymore?

A friend of mine who is the audio director of a church with a number of campuses spread around the city framed his situation for me. For them, if there’s a high enough volunteer to churchgoer ratio, then they’ll roster on a monitor engineer with a separate monitor console. For the smaller congregations with less requirements, they’ll typically run a mixture of pre-mixed sends and tablet app personal monitoring control. When I pressed him about potentially needing a more robust solution, he said there’s a hundred other things that are more likely to go wrong and more impactful than losing a wireless connection to a tablet app.

I was wondering if that was the end of it, if the scope of personal monitor mixing was now encompassed by those two scenarios. After all, I’d made use of tablet apps in the same way plenty of times and found them to be pretty useful. However, after just one go with Livemix, I can see why systems of its ilk still pop up onstage.

While tablet apps can technically perform the same core function as Livemix — i.e. crafting a personal monitoring balance — there are plenty of Livemix functions a tablet app can’t reproduce. All of which are due to Livemix being a standalone system.

I’ll paint the picture. When you set up a tablet mix system, it’s just another level of control over your main session. You set up an aux send, then let the performer control the levels of the channels feeding that send. They can’t — and shouldn’t — control the EQ, dynamics, or effect inserts of those channels. A standalone system like Livemix has no such constraints. On Livemix you can EQ each channel with a frequency sweepable, three-band parametric EQ and high-pass filter. You can also add basic compression with three levels of ‘squish’, and add effects. You can do all that on a perchannel basis or globally.

The other benefit of a standalone mix system operating outside of your main digital console session is the ability to save your individual

preferences. While you can technically save your last mix balance, it would take more effort than it’s worth to save multiple individual monitoring snapshots for particular songs inside the FOH session, or any level of personalisation that’s a doddle for Livemix.

Even though it was the first time he’d used a personal monitor mixer, the drummer was already crafting per song mixes in rehearsal, which would have been a nightmare to snapshot.

Livemix also allows experienced users to control other performer’s units remotely via the Mirrormix function. When activated, it turns the screen orange and lets you help them get in the ballpark and assign groups.

BREAKING MIX INTO COMPONENTS

The Livemix system hub is the Central Mixer/ Distributor. You can either plug a Dante option card into the back to feed it 24 channels of audio, or get the Livemix AD-24 24-channel A/D converter which feeds the Mix-16 Central Mixer with a single Cat5 cable. I was operating on a Digico SD9 with MADI outputs, so I opted for the analogue version. As well as ¼-inch TRS inputs, it’s also got DB25 connectors on the rear, so it’s easy to hook up looms of eight inputs from your stagebox.

Dante is a good choice, though it’d also be nice to see a MADI option card. Thankfully, both the analogue input and Dante options are the same price, so you only lose on the cabling side if you don’t have a Dante-enabled console.

The Central Mixer comes in two sizes; 16 channels and 32 channels. They only have eight and 16 ethernet outputs on them respectively, but here’s the kicker, each CS-Duo Personal Mixer — the mixer unit each performer manages — has two distinct mixers onboard, A and B, with their own main controls.

You only have to feed one ethernet cable to each CS-Duo, which carries all 24 channels and power for the unit. You can use any flavour of Cat5 or Cat6, but there’s no Ethercon receptacle on either end, which seems a bit odd for something onstage and priced as a pro product.

Besides that, the CS-Duo is professionally built.

Everything else about it seems quality. It’s rugged yet lightweight enough so it won’t make your stand unstable. I secured the clamp to both thicker music AT 45

stands and thinner microphone stands. You can also screw it onto the top of a spare mic stand if you have one. It has three main continuous knobs for each mixer — a master volume, dedicated Me control, and a volume/pan knob that controls most other functions — that feel solid to the touch and provide a nice range of motion. I did notice one unit’s ‘press to select’ function, which alternates the control between volume and pan, required slightly more determined pushes than its counterparts.

Aside from the three encoders, there’s also a Control button to nominate which side you’re currently adjusting, and also doubles as an intercom button when held, plus 24 backlit push buttons to select channels. While the push buttons are handy, in practise you’re more likely to select functions from the full-colour central touchscreen.

It’s resistive, rather than capacitive, which other than requiring a little more pressure is a far more reliable option. You can even use it with gloves on.

Each side of the mixer is colour-coded, blue for A and red for B, with the backlighting following your selection. Likewise, the screen goes red or blue depending on which side you’re controlling. If there’s only one performer per unit, you can alter it so both sets of knobs only control one side. Suffice to say, short of an AI bot slapping your hand, Livemix makes every effort to make sure you don’t alter the wrong mix.

You can also plug in the optional FP-2 twobutton footswitch to control channel and Me group levels — tap to level up and hold to turn down — as well as hold both down to activate the intercom.

The last piece of hardware is the DA-816 converter. It reverses the flow, converting personal monitor mixes back from digital to eight pairs of stereo TRS outputs. You only have to feed it one Cat5 cable for all those pairs, and it’s easily assignable via the CS-Duo. It’s built specifically to interface with racks of wireless in-ear monitor transmitters, rather than running analogue cables from the CS-Duos themselves.

SOUNDING OUT

The entire system operates at 24-bit/48k, and there’s nothing to complain about in terms of sound quality. The systems are quiet and have oodles of gain, enough that I had to limit the output to prevent potential hearing damage for in-ear users.

The 1/4-inch and 1/8-inch headphone jack are linked. When you select stereo headphone output on the unit, both physical outputs are mirrored, and when you choose mono balanced, it disables the headphone output. Presumably so you don’t accidentally plug a loudspeaker into the headphone AT 46

output, or vice versa.

Having an effects and EQ/ dynamics section separate to the main mix is also helpful.

Though anything other than mild reverb can be a hinderance when vocalists attempt to talk to the audience; using the ambient mic was a more natural option.

The two main issues I had with the units were to do with the Me group and Intercom.

When faced with the rising tide of stage noise, it got me thinking about other implementations of the More Me concept. There’s two ways you can go about it. The first, and Livemix’s method, is to have it function as a normal group. You assign one or more channels to it, and turning the Me knob changes the level of those channels. The second method is using it as a balancing control, that is, as you turn up your Me group it also turns the rest of the channels down. I prefer this method for personal monitor mixing, as it does what most performers are really trying to achieve — poke their sound above everyone else — without increasing the overall stage volume. It’s not just for wedges, it’s a handy function for in-ears too.

The other feature I’d like to see implemented is assignable intercom mic channels. The intercom mic channel uses the same microphone as the ambient mic channel, it just momentarily boosts and compresses the signal. While the ambient mic is really handy for dialling in a bit of liveliness into your mix, I personally didn’t find the intercom that useful in a live scenario.

Because it’s on the unit, it’s always at least arms length away from your mouth. With a standard shout mic, it’s the proximity and signal-to-stagenoise ratio that makes them effective. If you want to communicate mid-set with other performers or engineer, you have to actually shout over the noise to be heard — not a good look. It would have been useful to include the option for an intercom mic input, or be able to assign any input — which could be your SM58 shout mic — to the intercom channel.

The intercom is really more suited to practise situations. It could be really useful for communication within large ensembles, and is the perfect solution for tracking in the studio.

Livemix also has practises covered with its 1/8inch stereo aux input. You can plug a phone in and show everyone exactly how the middle eight should ‘really’ sound. Likewise, there’s an onboard metronome with a few set tempos and beat divisions which covers most bases. You can’t sync

it to an external playback source, but it’s useful for getting those chops dialled in before the gig.

PERSONAL CHOICE

If you’re in the market for personal monitor mixing, there’s still no better solution than a dedicated system. While tablet apps have certainly changed the game by making personal monitoring control more accessible, they still only do one thing — control.

Every CS-Duo has onboard audio features like the ambient mic, intercom, aux input, local headphone and balanced outputs. They’re more than just handy, they make each unit worth their weight. Throw in built-in effects and metronome, and the ability to save multiple monitor mixes separate to the console’s mix session and the value adds up.

Beyond the stage, a Livemix system would make a supercharged headphone system for any studio, letting you focus on engineering. The intercom mic system enhances communication dramatically over shouting into room mics, and the onboard ambient mics can help add a bit of liveliness into the performance.

There’s plenty to set Livemix apart from its competition too, specifically the high number of inputs, dual mixers per unit, and the full-colour touchscreen, which really makes operation a breeze. With a couple of firmware upgrades to enhance the unit’s flexibility a hair, the Livemix would have everything you could ask for, especially more Me.

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REVIEW

ARTURIA V COLLECTION 5 Virtual Keyboard Compilation All the numbers add up as Arturia’s fifth V Collection grows with five new instruments.

NEED TO KNOW

Review: Preshan John

PRICE Expect to pay $599 Upgrade: $259 CONTACT CMI: (03) 9315 2244 or sales@cmi.com.au

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PROS A great collection gets bigger Resizable windows Analog Lab gets smarter

CONS None

SUMMARY Arturia already had an incredibly strong lineup with its V Collection, so it’s surprising how valuable the new additions are to any soft synth enthusiast. Rhodes, B3, Farfisa and Synclavier aren’t names dug out of obscurity, and the fifth ‘instrument’ is actually a whole collection of pianos in itself. As always, worth it.

Arturia’s V Collection is a virtual candy store for vintage synth enthusiasts. They’ve been used on countless records as stand-ins for the often unattainable ‘real deal’, and the collection keeps growing. The fifth instalment has five new keyboards added to the mix, had its main menu system overhauled, and, would you believe it, now features resizable windows from 60-200% — no need to squint any longer!

Before we get into the new bits of gear, let’s recap and see what else Arturia has been fiddling with. Three of the newer additions — Matrix 12V, Solina V and Vox Continental V — remain unchanged other than the zoom factor. Others have had cosmetic changes, like the ARP 2600V, which has shed its Blue Marvin skin for the 2601 orange and black colour scheme. Oddly, the Jupiter 8V has moved away from the original hardware colour scheme for a completely new look.

Others have been rearranged. The Modular V’s floppy patch cable physics have been tightened up and the modules restacked with filters at the top and sequencer closer to the other controllers. Both the CS-80V and Oberheim SEM V have gone a little off script. Arturia pulled out the CS-80V’s secret door modulation assignment panel and put it ‘under the bonnet’, leaving more room on the surface to spread out those faders. There’s also no more annoying rotating fans behind the grille. The Oberheim SEM V’s arpeggiator, tune and portamento functions have been moved up into the top panel, adding pitch and mod wheels next to the wider keyboard. Arturia is obviously shooting for better usability rather than true-to-life panel layouts, which is a welcome change.

The previous version looked like you were squinting at a Wurlitzer through a letterbox slot, while the new one lets you see its legs. Underneath it also adds a five-slot pedal board with 11 effects to choose from, and an external LARC-looking reverb control.

The Prophet V is much snappier when flipping between the 5, VS and hybrid engines — they slide across rather than unfold excruciatingly slowly. The VS engine is much clearer now, with the modulation, wavetables, envelopes, etc, now all selectable via tabs, while the straight-up Prophet V has additional chorus and analogue delay pedals. Likewise, as well as getting nicer-looking wooden end cheeks, the Mini V’s additional modulation/arp/ motion flip-out screen looks much more modern.

Analog Lab has undergone the biggest makeover. The emphasis is on search, with a new search input field and keyword browsing. It’s also de-emphasised the distinction between Single and Multi-engine sounds. Your choices feed into a new mixer, which is tabbed rather than scrolled, though it still provides the same functions — level control, MIDI assign, effects and preset management. Now, when you edit the preset, the instrument appears in the top panel rather than in a separate window. One great update is the ability to have separate lots of mapped controls for each of the two multi-patch parts, previously you had one global set of controls for a Multi-patch. Also gone are the Chords and Snapshots related to Arturia’s Keylab hardware. You can always assign chords to buttons via Arturia’s MIDI control centre, and the new Playlist feature is a much easier way of building sets without the limitation of only having 10 available spots.

That’s enough about the old synths, here’s a deeper look into the new babies of the bunch.

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PIANO V: HAMMERED HOME

The piano is my first love, so I immediatey gravitated to Piano V. Delightfully rich piano sounds pour out of this collection: the Concert Grand and Classical Upright no-frills emulations are bang on the money and addictively satisfying to play, as are the more specialised Intimate Grand, Jazz Upright, and Piano-bar Upright models.

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Among the controls, which include String Tension, Stretch Tuning and Unison Detune, I found the hammers’ Hardness setting to be particularly useful. It does exactly what it implies, altering the firmness of the attack, giving you an organic and natural way to balance the tone and feel of each note. In addition to the well executed standard piano tones are the more left-of-centre emulations like Glass Grand (which sound like: honky-tonk piano meets DX7) and Metal Grand (dry and attack-y).

In the Mix section of the GUI, you can play around with the mic setup, room setup, and a three-band master EQ. Upright piano models offer a choice of AB, Decca and Hybrid mic positions and a mixer lets you balance the mic levels. It would’ve been nice to see M/S as an option for the Grands, but the four choices of different AB and XY configs provide good variation in the piano’s spatial presentation. Room Setup places the piano anywhere from an Abandoned Warehouse to a Home Studio to a Wooden Room. Most are very usable and pleasing.

There are truer options if you’re searching for the ultimate virtual acoustic piano sound but, for what you’re getting in V Collection, Piano V is an outstanding newcomer to the pack that won’t disappoint.

SYNCLAVIER V: RETURN OF THE BEAST

Of the five new models in V Collection 5, Synclavier V is the only pure-blood synth. It’s an enormously powerful instrument, with the combination of FM and additive synthesis offering an overwhelming breadth of tonal possibilities.

Even after tinkering with it for hours, I felt like I’d only scratched the surface.

Firstly, Synclavier V’s good looks deserve a moment of appreciation. It’s almost as nice to behold as it is to play. Global controls are located directly above the keyboard section — things like Timbre Settings, Amp Envelope Offset, Harmonic Envelope Offset, Arpeggio, Polyphonic Mode and Portamento. Click the arrows to reveal the next module where you can have more in-depth control over the many layers that contribute to the Synclavier sound.

Open up the SCR tab and the GUI changes to a sci-fi-esque nerd lab. This is the guts of the machine where you can activate up to 12 Partials and dial in your selected amplitude and harmonic envelope for each, with the help of a graphical display. The Time Slices tab allows precise harmonic construction of each Partial, and the Mixer tab lets you balance the levels of the Partials.

The FX section gives you all the usual suspects — reverb, delay, flanger, phaser, chorus — and you can apply up to two simultaneously.

Synclavier V is capable of glistening pad sounds, percussive stabs, sub-bass rumble, horn sections, orchestral strings, techno leads, and outright wacky convulsions. The comprehensive preset list is a great place to start your Synclavier journey, and Arturia’s categoric folder structure helps you quickly get in the ballpark of the tone you’re scouting for.

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FARFISA V: MAJOR ORGANS

The Farfisa V is having an identity crisis. It looks like an organ, but at heart, it’s an all-out synth.

While it’s capable of great organ tones, don’t let that stop you from going wild with the knobs and sliders to create sounds that don’t even vaguely resemble a traditional organ.

All the customary controls are at your disposal.

Start out by determining the character of your sound with white voice switches, then tweak it further with the reverb, percussion and tremolo options.

As with most of V Collection’s synth modules, the real fun starts when you open up the top panel.

Here you can flick a switch to engage user-drawn additive waveforms, allowing you to create some extraordinarily complex harmonic tones by simply mousing in patterns. Attack and Release settings let you shape the envelope from a hard-hitting lead tone to something more pad-like. The Voice Tune knobs let you fine tune the overtones.

Thanks to the split keyboard you can assign the lower octaves (coloured grey and black) to a bass sound. The five stomp box effects (flanger, phaser, chorus, analogue delay and overdrive) sound good and are well-suited to the general character of the Farfisa. Using these effects to heavy-handedly mangle the tones resulted in some very unique and satisfying textures. There are several reverb options to place your tone against the right aural backdrop, from cavernous halls to wobbly springs. Basic three-band EQ appears on both the organ and the speaker, along with the option for on- or off-axis mic placement.

STAGE-73 V: OPEN RHODES

Detail and articulation are what helps the Rhodesmodelled Stage-73 V stand out. From the crisp, delicate bell-like attack, to the woody, dry sustain of each note, I dare say this is about as much fun you’ll have next to playing the real thing.

The Stage and Suitcase models give you two primary sound variation starting points. Selecting the Stage model brings up a Fender amp with a bunch of settings of its own including a typical three-band/three-knob EQ, reverb, vibrato speed and intensity controls, and a switch for Bright and On-Axis. Under the bonnet you can refine the sound’s character with the Harmonic Profile menu options, and alter the percussiveness or smoothness of the decay with the Tone Bar Resonance knob.

As with Piano V, the Hammer Hardness control adjusts the aggression of the attack. Pair that with some compression and overdrive for a tone that growls on the low notes and sweetly saturates on the highs. You get a little inventory of stomp box effects, similar to Farfisa V, plus a compressor pedal. Dial in some Chorus or Phaser and your ears will thank you.

One of the Fender Rhodes’ signature traits is the way its character changes with velocity. Stage-73 V gives you complete control over this with the Velocity Curve graph, where you can literally plot how you’d like the tone to change — gradually, or with a more definite, predictable switch-point.

While Synclavier V is a behemoth that

encourages tonal adventure, Stage-73 V humbly recreates a single sound — and it does it well. You can’t deviate too far from the classic Rhodes tone, but every control on Stage-73 V feels purposeful and deliberate in securing just the right version of that tone you’re looking for.

B-3 V: ARMED WITH HAMMOND

The king of organs, Hammond’s B3, has always felt missing from the V Collection. Arturia knew it and the wait has been worth it. B-3 V’s upper and lower keyboards each have its own set of drawbars which can be key-switched to different positions using the notes of the first octave (inverted colours). The Leslie cabinet, modelled with grit and realism, also has settings for various types of reverb, rotary drum speed and acceleration, horn acceleration, and stereo width.

Although not quite as nimble as the Farfisa V, the B-3 V is still capable of a surprising range of tones that will have you playing a best-of from your LP collection. The stomp box effects — chorus, distortion, flanger, phaser and more — instantly stimulate the creative juices if you want your organ to go somewhere new.

Arturia has taken the liberty of adding synthlike features to B-3 V. Opening up what I like to call the ‘geek drawer’ reveals a more technical interface where you’re presented with three modulation options — a Multi-Point Envelope, LFO, and Step Sequencer. All parameters are based around the modulation of the drawbars, with the upper and lower sets displayed as the top and bottom of the graph. The 32-step sequencer lets you dial in patterns; the LFO cycles between your drawbar settings according to the selected wave type; and the Multi-Point Envelope gives more precise control over how a note morphs harmonically over time. This section takes a little nutting out, but kudos to Arturia for alighting upon a great way to add automated drawbar modulation that offers creative control without sacrificing its natural tone.

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REVIEW

KORG NANOKONTROL STUDIO Portable MIDI Control Surface Review: Preshan John

I’m usually not a big fan of non-motorised control surfaces. The constant dance of picking up last used settings drives me nuts. As a result, I’ve never really fallen into the orbit of these devices. Recently, I was given Korg’s new NanoKontrol Studio to review. Intrigued by the miniature box, I downed some humble pie and set out to discover if the little, non-motorised controller could soften my defiant stance.

NanoKontrol Studio is a mini control surface with eight channel strips, each featuring a (nonmotorised) fader, pan pot, and backlit solo/mute/ select/record buttons. To the left you get your usual transport controls, plus a jog wheel and marker setting buttons. Nothing about the unit jumps out as ‘special’, except perhaps its super-slim form factor.

PORT-ABILITY

NEED TO KNOW

Korg’s Nano family — NanoKey, NanoKontrol and NanoPad — are all nifty portable music production tools primarily targeting the music producer on the go, and NanoKontrol Studio really amps up the mobility stakes with Bluetooth compatibility. But first, how are those faders?

One way non-motorised faders work effectively is by implementing a method of picking up their last position by forcing you to return the fader to that position before the DAW fader responds to any more input. This is typically a DAW-specific parameter; in Logic and Ableton you have to set the controller operation to ‘Pickup’, Pro Tools was a little less helpful. It stops the DAW fader jumping to the physical fader position as soon as you move it.

PRICE Expect to pay $199

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

Even with the option turned on, it can get tedious having to constantly wiggle faders to get them going, and it also reduces your mix precision as you overshoot previous level settings. If you’re not using it for critical mix applications, it can be fine.

NanoKontrol Studio has a couple of other party tricks though. Firstly, it does more than just control your DAW — you can use it to control your iOS recording apps. Secondly, it has low-latency Bluetooth connectivity for wireless control. That’s pretty fresh.

The featherweight plastic unit feels slightly toyish but doesn’t come across as flimsy. The jog wheel is firmly detented and the rubber buttons look and feel great in use, with the backlights fading in and out with a classy glow. The faders and knobs aren’t exactly pro, but they’re comfortably smooth, with reasonable travel for the unit’s size.

The Power switch has three positions — USB, Standby, and Battery. If you leave it on Battery even when connected via USB, it’ll lose charge. But the unit has automatic power-off when used wirelessly.

Detailed instructions are provided to get it running with most common DAWs. If Pro Tools is your DAW of choice you’ll have to use NanoKontrol Studio in HUI emulation mode. You do this by pressing a combination of buttons while powering the unit on, and a rummage through the user manual elicited the NanoKontrol Studio’s secret HUI code (the ‘Scene’ and ‘Rewind’ buttons).

Also in HUI mode, the throw of the pan pots seemed to fall short of their corresponding value in Pro Tools — pan 100R looked more like three SUMMARY While the lack of fancier features leaves a little to be desired, the Korg NanoKontrol Studio is small, light, operates wirelessly, and is battery-powered. If that’s what you need in a control surface, the price tag sure isn’t going to stop you from snapping one up.

o’clock on the surface. Weird. Endless encoders may have been a neater option here. The faders’ resolution is in 0.4dB steps — sufficient for 90% of typical automation writing needs.

KONTROL EDITOR

Customised control comes by way of Korg’s Kontrol Editor software. This allows you to assign specific MIDI commands to all the controls (faders, buttons and knobs) available on the surface. GarageBand and Logic Pro X users get a plug-in which connects you directly to NanoKontrol Studio from within your DAW. Editor allows for up to five scenes, letting you reconfigure the entire surface with the click of a button, perfect if you’re using it with multiple applications.

Wireless Bluetooth control is, in some ways, NanoKontrol Studio’s saving grace. Pair this with custom configuration via Kontrol Editor and you can get rather creative with how it’s used. It’s handy being able to pass it to a drummer or guitarist with submix tracks mapped to the faders so they can adjust their headphone mix. Just be sure to ask them to leave the transport controls alone.

WOULD I?

Since the price of this thing is less than a power bill, it’s worth a serious look for the wireless functionality alone, provided your computer supports the Bluetooth chipset. It’s a little impractical for me, only because my main application is in-the-box mixing, and I’ve become accustomed to touch-sensitive, motorised controls. Nevertheless, there are plenty of scenarios where you don’t need all that fancy business — maybe it’s controlling your backing track levels on the live stage, or using it to keep a handle on your main DAW outputs, or giving musos wireless control of their headphone mix in the studio. In which case, NanoKontrol Studio might be the ideal fit for your needs.

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REGULARS

Last Word with

Michael Carnes, Part II Founder of Exponential Audio

Last issue, Michael talked about his time at Lexicon and how he started Exponential Audio, a developer of plug-in reverbs and effects. This issue he tackles the sticky debate of convolution vs algorithmic reverbs, and hardware vs software.

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The issue with convolution reverb is when you take an impulse response of a hall, you’re typically only projecting the impulse from a few places on the stage captured by microphones in only a handful of places. You will get a perfect impulse response from that point source to that microphone at that instant in time.

Of course, things change — people move around, humidity changes. Just say you had a perfect impulse response, I defy you to find an orchestra or band you can fit into two or three points. They occupy space, a thousand points of radiation would be understating it.

If you moved around a few seats from the microphone’s position, it would sound pretty much the same, whereas the impulse response would be different. What’s happening? The impulse response is a primary expression of what’s going on, but it’s not what you hear.

The human brain and hearing system is seriously into data reduction. Our brain could not possibly cope with hearing individual impulses. We’d have to eat 50 million calories a day just to get the energy into our brain to do it. You think MP3 is bad, a whole lot less gets to your brain from most of the things you hear.

It’s critical we hear early sounds to identify where something’s coming from because that might be something that will eat us. There’s no evolutionary advantage to having a more sophisticated sense of reverb.

Your sense of space is driven by the inter-aural difference; the momentary sensation of positive pressure on your left ear at a certain frequency, negative pressure on your right ear at the same frequency, and whether that changes 10ms later. A convolver could emulate that with enough impulse responses taken from enough points around the room with enough microphones. But why go to the bother? It’s a huge amount of computing for no advantage.

A lot of reverberation is about trying to get a sense of what you really hear, and what matters. It turns out you can express that with a lot less data and a lot less computing power. Which means you can run more reverbs in a mix because you’re not bogging down a processor.

In the end, you might want to see a picture of an Abbey Road chamber while you’re doing a mix, but if it doesn’t work in the mix, it doesn’t work. If you want to get a little more space around a bass, you don’t want to dive through impulse after impulse trying to match a picture of what you think it’s supposed to sound like just to tame it at 50Hz. It’s a lot easier to change a parameter. That’s the advantage of algorithmic reverbs, you can tailor them to your material a lot better.

Could I model a room that’s 12 miles across, four miles high, with linoleum surfaces? Yes, but it wouldn’t be useful in a mix. Some of my reverbs will give you ridiculously long reverb times, but it’s just something that falls out of the math rather than a pursued outcome. Some people will dream up

something really novel, so you give them a bit of rope to hang themselves with. A lot of the boundaries comes from thinking practically; you can’t be infinite.

I’ve been at it long enough that I know the rules fairly well.

I’m looking more for atypical stuff, which tends to be bad environments they might need in post — closets or bathrooms — or good exteriors.

My wife isn’t embarrassed any more by what I do. There’s hardly a canyon or hillside I walk by that I won’t clap in and make mental notes about what I heard. Halls are the same way; you shout, you sing, you clap your hands. If there’s some percussion instruments in there, you give them a whack if you don’t get caught.

Everyone in the reverb business is way past 90% good. Now we’re shooting for 95% good, 99% good, and so on. To make it absolutely clear there was ‘no artificial reverb used’.

I record a lot of chamber music. There’s no money in it, so a lot of the performance spaces are just awful — trucks going by, harsh art galleries. Most of my mic placement has more to do with eliminating the room than anything else. Once I’ve cleaned it up (kudos to Izotope) I’ll pop some of my own reverb on top to bring a sense of space. My favourite thing is when musicians say, ‘I love how you captured what I heard in that room,’ when there’s none of the original room in there.

Confirmation bias says whatever you believe is what the facts are going to support, no matter what they really say. An awful lot of people are invested in saying hardware reverb is better than software reverb. Firstly, hardware reverb is just little processors running software. People will argue, ‘there’s something about the converters.’ Yeah, there was, 25 years ago.

Even crap converters are really good now.

It partly boils down to workflow. They’re moving faders on a desk, have got a LARC, and that’s the way they’ve worked for 20-30 years. If they’re making good stuff, more power to them.

If you have an algorithm that demands the power of entire CPU, and had a double blind metric where people could agree that one algorithm is demonstrably better for a given style — because pop music demands a different vocabulary from classical or folk music — well, okay, that hardware might be better. A few years later, a software reverb will take care of that just fine.

Nevertheless, I’m unaware of any algorithms with that level of demand. I have a few friends in the business on the hardware side that insist they need that level of power and would find it very difficult to port to a computer. I’m not sure that’s the case, but they’re doing fine.

Any of my reverbs use considerably more power than any reverb Lexicon has ever made in its entire history.

However, I can take an eight-core ‘trashcan’ Mac and run a couple of hundred instances at once. That tells you how fast technological progress really goes. Why go back?

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