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The RØDECaster Pro is designed to simplify podcast production whilst delivering superb audio quality. It supports up to four presenters/guests, as well as offering easy connection to phone, USB and Bluetooth™ sources. Eight programmable pads offer instant playback of sound effects and jingles. Podcasts can be recorded directly to microSD™ card, or to a computer via USB. Ease of use is assured, with intuitive controls and large full-colour touchscreen.

The Cho oice off Toda ay’s Crea ative Ge ene erattion.™ AT 2


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

Regular Contributors Martin Walker Paul Tingen Brad Watts Greg Walker Andy Szikla Andrew Bencina Jason Hearn Greg Simmons Mark Woods Ewan McDonald Guy Harrison

Art Direction Dominic Carey dominic@alchemedia.com.au Graphic Designer Daniel Howard daniel@alchemedia.com.au Advertising Philip Spencer philip@alchemedia.com.au Accounts Jaedd Asthana jaedd@alchemedia.com.au Subscriptions Sophie Spencer subscriptions@alchemedia.com.au Proofreading Andrew Bencina

AudioTechnology magazine (ISSN 1440-2432) is published by Alchemedia Publishing Pty Ltd (ABN 34 074 431 628). Contact +61 3 5331 4949 info@alchemedia.com.au www.audiotechnology.com PO Box 295, Ballarat VIC 3353, Australia.

All material in this magazine is copyright Š 2019 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. 7/3/2019.

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COVER STORY

Universal Audio Apollo x8 Audio Interface

Leslie Braithwaite — Beyoncé, Migos, N.E.R.D. & Cardi B

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dB Technologies ViO L212 & S218 Launch

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Arturia Drumbrute Impact Drum Machine

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ISSUE 55 CONTENTS

Neumann, Surviving 90 Years of Success

50

20

Quick Mix, Sarah Maddigan 18

Allen & Heath SQ7 Digital Console

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THIS IS STATE-OF-THE-ART WIRELESS COMMUNICATION

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GENERAL NEWS

NEUMANN CANS! The Neumann NDH20 is a closed-back headphone designed for excellent isolation, making them a detailed and balanced monitoring and mixing headphone, even in loud and noisy environments. As high-isolation headphones, the NDH20s are best suited for monitoring in the tracking room as well as FOH work. Thanks to its extended frequency response ranging from 5Hz to 30kHz the NDH20s are particularly helpful to check the upper and lower extremes of the audio band. Neumann also cite the ‘unusually even frequency response’ and ‘natural stereo

image’ of the NDH20s, providing an open-back-like mixing experience on the road. They feature 38mm, 150 ohm dynamic drivers, peak sound pressure of 114dB, an adjustable and foldable headband, two detachable and replaceable cables, as well as replaceable circumaural memory foam earpads. Sennheiser: (02) 9910 6700 or sales@sennheiser.com.au

RODECASTER PRO GETS MULTITRACK RECORDING By way of a new firmware update 1.1.0, Rode Microphones has enhanced its RodeCaster Pro podcast production console with the highly requested multitrack recording feature. The update also brings a more user-friendly interface and channel selection. All existing customers will be able to update their RodeCaster Pro with multitrack recording, and all new units will feature the update. The multitrack feature will record 14 tracks; A stereo ‘live mix’ track, as featured since its release, a mono track for each of the four microphone inputs, and a stereo track each

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for the USB, 3.5mm TRRS, Bluetooth and sound pad channels. Multitrack recording can be activated via the Advanced settings in the Hardware section of the touchscreen interface. From there, a single switch engages ‘Multi-Channel’ mode for multitrack recording. The 1.1.0 firmware update will be available to download this month from rode.com/rodecasterapp. Rode Microphones: (02) 9648 5855 or info@rode.com

MORE NEWS AT www.audiotechnology.com.au


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ANTELOPE STEPS INTO EDGE GO Antelope Audio hit 2019 running, launching three new products: Edge Go, a bus-powered modelling microphone; and two new flagship audio interfaces Orion32+|Gen 3, and Orion32 HD|Gen 3. Edge Go is the new portable member of the Edge family of modelling mics. A USB-C connected device, Edge Go achieves an entire processed vocal chain at the hardware level all in real-time, monitor-able via 3.5mm jack. A variety of condenser mic models are available, along with vintage compression and EQs, as well as built-in de-essing, gating/ expansion, tape saturation, and reverb all controlled by a PC/

Mac app. The Orion32+|Gen 3 brings the same connectivity, flexibility, and channel counts to the table the original Orion32+ made its notable name with, but with upgraded operating levels (+24dBu max), upgraded AD/DA conversion, ultra-fast and stable custom Thunderbolt driver for Windows, and a major facelift. Orion HD|Gen 3 gets similar upgrades and facelift. Federal Audio: 0404 921 781 or sales@federalaudio.com.au

AUDIENT SONO: 12AX7 ONBOARD Audient has teamed up with cab simulation pioneers Two Notes Audio Engineering to present Sono, a new audio interface for guitarists. Combining Audient’s analogue and digital conversion recording technology with speaker-cab simulation from Two Notes, Sono houses some tweak-happy valve goodness. Featuring an onboard 12AX7 analogue valve and threeband Tone control alongside Two Notes Torpedo power amp modelling and cab simulation, Sono provides a whole bunch of tonal options for guitarists, whether they’re recording, practising

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or gigging. The interface features near-zero latency, a monitor mix feature for blending guitar input signal and DAW playback, as well as comprehensive I/O for recording more than just guitar: two Audient console mic pres and ADAT expandability to 10 inputs for drum recording. Studio Connections: www.studioconnections.com.au

MORE NEWS AT www.audiotechnology.com.au


Musos Corner Celebrating 50+ Years in the Music Industry

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PMC’S NEXT GEN IB2S PMC claims ‘More power, more detail and more flexibility’ for its new IB2S-AII and twin cabinet IB2S XBD-AII reference monitors. This new generation of PMC’s IB2S active range is aimed at professional users working in stereo and surround formats for reference and mastering applications. New in these models are 3U rackmounted electronics, one per channel, delivering DSP-controlled Class-D amplification from afar — 2025W to the single cab model and a hefty 3225W to the latter. PMC believes this increase in power will diminish distortion

and gives greater bass definition, attack and extended headroom. Both systems also feature HF and LF shelving filters, ±8dB input level trim, and an AES3 digital input. The input sensitivity of the balanced analogue inputs can also be adjusted from +4dB to +16dB, all of which can be accessed via an RJ45-connected tabletop remote. Interdyn: (03) 9426 3600 or sales@interdyn.com.au

TASCAM INTRODUCES DR-X SERIES Tascam introduces the DR-X Series, the next generation of its line of professional grade handheld recorders. The DR-40X features integrated unidirectional stereo mics with scalable A/B or X/Y configuration, dual XLR/quarter-inch combo inputs, built-in phantom power for condenser mics, integrated four-track capability, and wired remote control option, as well as Tascam’s Auto-Tone function, providing an audio cue tone identifying each recording take. The DR-40X will be joined by the affordable DR-07X, which, with the same mic configuration, Tascam claims will deliver ‘professional performance’ for

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musicians and voiceover artists. Finally, the DR-05X is equipped with a pair of omnidirectional condenser mics, making it better suited to recording music, meetings and dictation. Each unit incorporates a 2-in/2-out USB audio interface, microSDXC storage up to 128GB and a daylight-viewable white backlit display, along with an Overwrite mode with one level of undo. CMI Music & Audio: (03) 9315 2244 or www.cmi.com.au

MORE NEWS AT www.audiotechnology.com.au


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LIVE NEWS

EAW: FIRST PA FOR NEW ERA EAW has launched the new KF810P line array. For the moment, it’s designed specifically for installed applications, but you can be assured there will be a touring version down the pipeline. The KF810P integrates two high-powered, ported, 10-inch low-frequency drivers with three-inch voice coils; four fiveinch midrange drivers with 1.7-inch voice coils; and two high-frequency drivers with 1.4-inch exits and 3-inch voice coils. Tuned, phase-aligned spacing of the low-frequency components extend pattern control. The output of these sources unites through an integrated horn that occupies most of the

forward face of the enclosure, delivering up to 145dB SPL with accurate pattern control down to 250Hz to master challenging acoustic spaces. EAW claims their Concentric Summation Array technology results in ‘smooth, accurate, studio monitor-quality response’ from 50Hz to 20kHz. Nominal beamwidth is either 80° or 100° horizontal by 10° vertical. PAVT: (03) 9264 8000 or www.pavt.com.au

ADAMSON CS7P: MILAN READY Adamson introduces the CS7p point source enclosure, the first of the new CS-series of mobile loudspeakers featuring Class-D amplification, DSP, and Milan-ready (AVB) network endpoints. The CS7p has already been successfully used as a FOH nearfield monitor at Hillsong Conference (pictured) and on the Drake and Migos North American tour. “In an application such as FOH monitoring, Milan simplifies the deployment process by allowing users to make a single connection,” says Benoit Cabot, Director of R&D at Adamson. One of the benefits of Milan is being able to daisy-chain networked audio between multiple sources in

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addition to an analogue XLR input and output. Acoustically, the CS7p contains two seven-inch Kevlar Neodymium transducers and a three-inch compression driver, loaded with a rotatable 70° x 40° (H x V) waveguide. The dipole arrangement of the cabinet provides a stable polar response, and the CS7p can also be paired to increase horizontal coverage and overall output. CMI Music & Audio: (03) 9315 2244 or cmi.com.au

MORE NEWS AT www.audiotechnology.com.au


Designed for Excellence in Every Detail. Introducing Apollo X. UA’s new flagship interfaces, featuring elite-class A/D and D/A conversion, new HEXA Core UAD plug-in processing, Unison preamps, +24 dBu operation, Surround Sound,* and more. All in a sleek Thunderbolt 3 package for Mac and PC.

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Distributed Australia wide by CMI Music & Audio, find your nearest authorised dealer at cmi.com.au AT 15


SOFTWARE NEWS

ABLETON LIVE: POINT ONE Live 10.1 is a free update for all Live 10 owners. New features include: User Wavetables, expanding the capabilities of Wavetable’s oscillator section; and Channel EQ, a flexible and simple EQ with curves and gain ranges suitable for a variety of audio material. The shape of the filters adapt based on how controls are set to ‘always provide musical results’. Also added to 10.1 is Delay, which combines Simple Delay and Ping Pong Delay and adds feature upgrades. Ping pong behaviour, as well as Jump, Fade-In and Pitch controls from those devices are all accessed from the front panel. Users will also now be able to

choose from a palette of automation shapes, stretch and skew automation and generally have greater flexibility in automation editing. Also added will be track freezing with sidechains, VST3 support and single track export with return and master effects applied. CMI Music & Audio: (03) 9315 2244 or www.cmi.com.au

IK MULTIMEDIA SAMPLETANK 4 IK Multimedia’s SampleTank 4 is the next generation of its sound and groove workstation offering pro-quality sounds, an intuitive interface and powerful editing and effects. SampleTank 4 offers a greatly expanded sample library (up to 8000 sounds across 260GB of samples) and a completely redesigned, scalable/ zoomable interface. A new sound engine offers efficient harddisk streaming. The library and sound engine are complemented by dynamic groove ‘players’ (arpeggiator, strummer, pattern and loop player), 13 new effects (for a total of 70), a powerful new Mix window, a unique Live mode and integration with IK’s iRig

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Keys I/O line. Three versions are now available: SampleTank 4 SE, SampleTank 4, and SampleTank 4 MAX. A new Mix window has been added to adjust volume, panning, mute, solo and routing for all 16 tracks simultaneously. A unique, re-designed Live mode offers fingertip-ready controls for live performance and lets you build an entire set list and create song presets. Sound & Music: (03) 9555 8081 or www.sound-music.com

MORE NEWS AT www.audiotechnology.com.au


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QUICK MIX

The

with

Sarah Maddigan

Who are you currently touring with/mixing?

I’m currently touring with The Preatures and The Veronicas as their monitor engineer, and I mix FOH for Jen Cloher. I also work at Northcote Social Club as an in-house production tech and freelance FOH/monitors/stage tech for a few production companies. What are some other acts/bands you’ve worked with?

The list is very long: Tkay Maidza, George Maple, SAFIA, Abbe May, LANKS to name a few. Courtney Barnett and all of the amazing Milk! Records family are regular clients and friends. I also have production managed multiple venues across Melbourne including Shebeen and The Curtin Bandroom. How long have you been doing live sound and what was your path to a career in audio engineering?

2018 marks my tenth year working in live audio. I started out as a teen, when my music teacher taught me how to setup a PA. I ended up studying at SAE because I didn’t want to study music at uni and it turned out I was pretty good at mixing. I started mixing friend’s bands, who then networked me with their friend’s bands and it all snowballed fairly quickly from there. Now I’m working five to six nights a week all over the country. What is your favourite console and why?

I’ve always been a fangirl of Digico; for sound quality, ease of use and flexibility. Digico is always at the top of my production rider. The SD10 has been my favourite console for a long time. However, I’m definitely keen to get my hands on an SD12. Favourite microphone or any other piece of kit?

I don’t really have a favourite microphone. 90% of my personal live kit is Sennheiser mics, with a couple of Audio-Technicas, a single Audix and Shure. I definitely use my Shure Beta 91A the most out of everything I own. It’s super versatile for miking things from kick drums to grand pianos, to tap dancing boards (which I have had to do more than once). It’s also slim enough to get inside all kick drums with a port. It’s usually the only mic I will throw into my bag if I’m not carrying a mic kit. I’m also a big fan of my Audio-Technica ATM450s. They sound great on everything. Most memorable gig or career highlight?

Unfortunately, my most memorable gigs are the gigs where something has gone disastrously wrong. I guess you remember those AT 18

ones the most, so you remember what not to do or how to fix it. The most recent was at a festival where we had a catastrophic generator fail and lost all onstage power multiple times. Every time it happened I would have to power cycle my console and it would take me a couple of minutes to get back up and running again. Each time I’d got back to the point where the band could safely put their IEMs back in, we’d lose power again. A career highlight has to be the run of stadium shows I’ve been out on this year as a monitor engineer. It’s an entirely different level of production and the connections I’ve made and the amount of new things I’ve learned are definitely invaluable to me. Gig highlight for this year would probably have to be a short three-day run with Jen Cloher. I went from an analogue Soundcraft in an old cinema, which was a lot of fun, to a surprising community hall which turned out to be one of the best sounding rooms I’ve ever mixed in, to a sold out Croxton with four of my favourite musicians. What are three mixing techniques you regularly employ?

One thing I do with The Preatures is setup an SM57 a few inches over the kick drum and heavily compress it. Izzi prefers the sound of ‘old school’ drums, so rather than send her the close mic’d drums which sound very modern, she only gets that 57 and some OHs. Depending on who I’m mixing, I often use parallel compression, particularly on the drums. Back in analogue days this was simple to do, but nowadays a lot of digital desks have made it harder with latency and phasing issues becoming a problem. A way I’ve worked around this is to mix to groups and have duplicate groups for whatever you want to parallel compress. One group is uncompressed and the second is your parallel. However, this doesn’t work for all digital consoles. When I’m mixing monitors and FOH at the same time, if I have the channels available I digitally Y-split my vocal channels. I do this particularly with quiet singers. It gives me the ability to do different EQ and processing on each channel and get as much gain before feedback in both the monitors and FOH without compromising on audio quality.


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bodies and hold any excess cable while having the packs almost entirely hidden from view. That being said, I’ve been researching Waves LV1 a lot recently and it looks to be something that has a lot of potential. It’s something modular and interchangeable, so you can only fly with what you need if you need to save on weight. Plug-ins have definitely been considered a game changer for a lot of engineers, including me. Waves is by far the most popular and they’ve recently released some cool plug-ins I’m really keen to try out, including one called X-FDBK. It’s basically an automatic feedback analyser/suppressor. It would be handy for festivals and shows where you don’t have a lot of time to tune wedges. What are three pieces of gear or features that have been game changers for you?

The Digico SD11i has definitely been the biggest game changer for me. I tour one with almost all my artists. It’s convenient having a powerful console that is compact enough to fly around with you. It ensures the show is always going to be consistent and the best it can be. These are nothing new, but Ursa straps have been a huge help for me on shows. I work with numerous female artists who use IEMs and wear a lot of elaborate high fashion outfits on stage. Most of the time there is no suitable place for me to clip the pack onto their clothes. Before using Ursa straps, we would have to tape the receivers directly to their body, often with gaff as no other tape would hold up for the entire set. It’s definitely not something I recommend doing too often. Having Ursa straps, means I can comfortably and safely strap IEM receivers to their

How have your working methods changed since you began live sound mixing?

When I started out, everywhere I was mixing had an analogue console. Very few venues had digital consoles installed. They started changing over around the same time I was becoming super confident in my mixing, so I basically had to relearn my approach to mixing on a digital console. I used to have to take photos of my GEQs for each venue and photos of the console after the headliner had sound-checked so it could be recalled before they went on. Nowadays, instead of photos I have show files for almost every console, or can at least build half a show at home before turning up to the venue, then just save a full soundcheck. No more manual recalling. Any tips/words of wisdom for someone starting out?

Don’t be afraid to ask questions.

REVIEW

KORG ELECTRIBE WAVE iOS Music Production App Review: Preshan John

iPad apps are incredible value for money. At just over $30, Korg’s Electribe Wave iOS app isn’t far short of a fully-fledged beat production tool. With 16 tracks on offer, it can create polished tunes that sound way more expensive than the app’s price. The layout is divided into five tabs — Mixer, Sound, Sequence, Motion and Utility. With Synth and Drum instrument divisions within the Sound tab that flip the layout between keys or drum pads. Synth selections are split up into a PCM or Wavetable menu. While there are plenty of ways to tweak a sound, synth connoisseurs may be disappointed. For example, there’s only one envelope generator and it affects both the filter and amp. The two LFOs come in a variety of shapes, can be tied to a number of parameters and easily locked to the song tempo. The effects sound reasonably good. Getting started is a doddle with scale choices reflected in the keyboard layout. Hit Chord mode, engage the arpeggiator, spin an LFO, throw on some reverb, and you’re off. It doesn’t take long to generate complex and lush synth tones. Pure, iPadpoking fun. AT 20

$30.99 | www.korg.com

I expected MIDI editing a seven-inch iPad screen to be a nightmare. Thankfully, Korg makes it very intuitive. The approach takes a little adjusting to but once you ‘get it’ you can drop in notes, build chords and change expressive parameters with ease. Electribe Wave’s drum sounds were close to what I’d expect in a third party producer’s sample pack. It doesn’t sound cheap. You get eight drum tracks and a drop-down menu on each track lets you swiftly build your kit of choice. Sequence beats or perform them in Record mode, then hit the Sequence tab to dial in expressive parameters — rolls for single beats, MIDI velocity, the usual stuff. The Groove setting is a fantastic value-add. If you’ve sequenced a straight 16th-note hat pattern, choosing a ‘groove’ for that

track will alter the velocity of the hits; introducing movement without manual intervention. There’s plenty of storage for patterns and you can launch patterns in real time like Ableton or Bitwig’s clip launchers. It could turn the app into a backing track platform for a musician or singer on stage. You can easily save patterns to another slot as a starting point for the next section of your song. Automation is accessed via the Motion tab where you can insert real-time tweaks over a long list of parameters. From the user experience to the song-creating potential, Korg’s app implementation is top notch and I’d highly recommend checking it out — even if it’s just to stretch those musical muscles on your sofa after work.


WIRED

or WIRELESS Premium Dynamics or Premium Condensers Electro-Voice ND (76, 86, 96) premium dynamic and the new RE (420, 520) premium condenser handheld microphones are available in wired or wireless models. Five wireless capsules are available for use with the handheld transmitter and the new RE3 UHF wireless system. A bodypack system and several capsule options for instrument and lavalier/headworn mic applications are also available.

Ph: 1300 026 724

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Fax: (02) 8850-2230

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Email: boschcomms@au.bosch.com

| www.boschcommunications.com.au AT 21


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How Neumann plans to maintain its edge after 90 years of success, from the president himself, Wolfgang Fraissinet. Oh, and a new product scoop! Interview: Christopher Holder

Neumann isn’t celebrating 90 years in the biz by a string of lucky coincidences, assures president of Neumann, Wolfgang Fraissinet. Throughout its long history, the German microphone manufacturer has been delivering on one key proposition: riding at the edge of physics to help turn natural sound into pieces of art. That’s not going to change; but Neumann, as a company, must. Recently, we caught up with Fraissinet during his Australian visit to talk about the future. AT: What’s 90 years of Neumann really mean for the company? Wolfgang Fraissinet: The 90-year anniversary provides a good opportunity to talk about our history and even more importantly, talk about where we are going. Being 90, there’s a danger of being viewed as a little bit dusty and old — almost 100 years old. So I get the chance to talk about what’s going on within our research and development department where we gear up for the years to come. AT: Neumann has been a Sennheiser company for 25 or more years now. How much of that R&D happens at Neumann versus within the Sennheiser mothership? WF: R&D is within the Neumann headquarters, located in Berlin. Sennheiser R&D is in Hanover. We do of course have some synergies between both groups of engineers. Take, for example, the Neumann capsules for Sennheiser wireless transmitters. In that case it makes sense to give the best of both brands to the same customer to get better audio results. But other than that we are developing completely different tools and technologies at Neumann. We share information, we talk openly about our R&D with Sennheiser, but we owe it to our customers to remain as Neumann. AT: Neumann has recently reissued the classic U67. It feels like a brave move to bring back such an iconic microphone. The level of scrutiny is unlike any new microphone release. WF: Interestingly, we reissued the U67 in the early ’90s for a period. At that time we had enough parts from the original production of the U67 [1960-1971] which we used to assemble microphones. After that we stopped and we thought, ‘well, we can never do that again because we’ve used up all the parts’. We had some happy customers and no quality complaints — after all, these were U67s through and through. More than 20 years later we asked ourselves, ‘Can we recreate these parts, while conforming to modern RoHS rules?’ ‘Will the result really be a U67 so that we can very clearly call it a U67 reissue?’ ‘Or is it only close to the U67?’ If the latter was the case we wouldn’t have followed through and built it. The answer; a clear ‘yes’, it really could.

We have had customers comparing the 2018 reissue with originals from the ’70s and the ’90s — something we knew would inevitably happen. These frequency-by-frequency ‘golden ears’ tests reveal one difference: age. When you compare a brand new U67 with its brand new capsule, it doesn’t sound identical to an early ’90s U67 because of the 25 years of dust on the membrane. Other than that, when you see the circuit board you see the electronic components we’re using, the parts, which are more critical than the capsule itself, is all U67. Which is why we can call it a U67. AT: Part of the reason for the reissue must be to stop customers buying expensive clones? WF: If people spend seven thousand US dollars on a new U67 they’ve got the Neumann guarantee for spare parts availability, for maintenance and for everything you need for a professional studio tool. Buying an old mic is more of a lottery. And if you buy a fake then you can be even less sure. It’s easy to copy the shape of a Neumann microphone, copying the content is a little more tricky. AT: How vigilant do you need to be in protecting your intellectual property? WF: We pay close attention to people misusing Neumann designs and our intellectual property. If a product is called WA87, or something similar, it’s no coincidence. We will buy the product, test it, disassemble it and we see how close it is to a Neumann, from a non-professional or home recording customer’s point of view. If it’s almost like the real thing we will take people to court. Recently we had just such a product taken from the market and they were forced to pay a fine because they violated Neumann intellectual property. We don’t do this with every product that comes along looking like a U87. If it’s a cheap knock off then every customer knows the difference. AT: There have been some lovingly engineered U47 ‘clones’ produced over the years, for example. WF: Neumann is not an aggressive vendor of merchandise or an aggressive advertiser of our brand. After 90 years, we’ve learnt how to position ourselves within the market. If a product comes into our landscape — fakes, copies, close-to products and so on — we will do our best to talk to these people. Normally they’re industry colleagues, not people we want to sue. In most cases we find other ways to harmonise the landscape. But of course when people are really clearly violating our rights then we have to take action and we do that from time to time. We have to do it from time to time because otherwise even a court would say, if you don’t pay attention to your IP then anybody can copy your design and get away with it. AT: So what’s next for Neumann? WF: Neumann has to acknowledge the fact the

audio industry won’t remain a separate industry among future multimedia businesses — along with video and IT. We’ve noticed that audio and video is being increasingly integrated into IT systems infrastructure. For example, a live Olympic games broadcast will be controlled from a central headquarters and be more of a vast network of IP addresses than a traditional broadcast studio. AT: So what are you doing to address that market? WF: That’s something I cannot tell you. That’s what we’re working on now. But I’m not talking about the next microphone and not even talking about the next hardware

It’s easy to copy the shape of a Neumann microphone, copying the content is a little more tricky

component. I’m talking about completely different business models that include services and more, in conjunction with IT infrastructure and software engineering. At this point in time we are expanding our Neumann software engineering capacity. AT: Sounds like a major shift. WF: It’s our job to anticipate the future of audio. What’s more we also have to anticipate the business model behind it. We have to do that now. The changes are happening rapidly. The last three years has changed with a higher speed than five years ago. A new home recording market is growing fast. People are creating content in non-traditional ways. For these content creators, it’s a numbers game: they create hundreds of recordings and cut the best bits together. They don’t have the knowledge to make a perfect recording in their first, second or third try. But the results are good and they have hundreds of thousands of YouTube followers. It has nothing to do with the way we have been defining professional audio in the last 90 years but it has definitely a lot to do with the way we have to redefine professional audio into the future. AT: Podcasters are unlikely to know or care about the name Neumann. How does Neumann still command that ‘superior’ point of difference in markets it’s new to? AT 23


We could be talking about Neumann offering something for considerably less money… but it wouldn’t mean Neumann is a cheap brand

WF: What I’m trying to explain here without talking too much ‘out of our laps’, is we are not intending on let our existing customer base go, and say ‘well, professional products are too expensive, too complicated, too analogue’ and depart from that and do something different in order to serve the markets I just described in conjunction with IT, video and audio. Keeping faith with our existing professional high-profile customer base is non-negotiable. As long as people are asking for U87s, they will get them in the exact same quality as we are doing them today and have done for 45 years. But at the same time we’ve got to expand our business to new customer groups without violating our relationship with our existing customer. We could be talking about Neumann offering something for considerably less money compared to what people are used to from our large diaphragm microphones, but it wouldn’t mean Neumann is a cheap brand. That’s not what we want to do. But at the same time we don’t want to overcharge people just because of our nice brand. AT: Speaking of software, what does Neumann think about mic modelling? WF: It is something we have been talking about with a lot of people internally and externally since the Antares mic modeller was released. AT: They’ve come a long way since then. WF: Yes, I know and I’m not saying anything bad about any competitor or product, I’m just saying I had my doubts and I still do. When somebody says they have a machine or software that allows me to record something with an SM58, push the U67 button and suddenly I get a 3000 Euro recording out of a 120 Euro microphone… I have my doubts. A lot can be done in software and for an untrained ear the results may be of a good enough quality. For the customer base who is buying U87s today — working professionals — it is not even a discussion because they know the differences; which are not only explainable by talking about technical data, like SPL and self noise level. But it’s when you’re at the edge of physics, which we are as a manufacturer of hardware, and being used by people turning natural sound into something AT 24

which is a piece of art, a song or a speech or whatever. You cannot measure everything. Engineers want to measure everything but you cannot measure a recording and say it ‘sounds good’. There’s something unquantifiable going on. AT: What does Neumann think about how highend outboard has been successfully modelled. Have high-quality modelled analogue circuits severely cannibalised outboard sales? WF: There has to be a certain amount of cannibalisation — these products do have their customer group and there’s a need for these products otherwise people wouldn’t buy them. The better the software modelling, the better the competition. From a Neumann perspective, high level competition is a good thing. Maybe some of that competition also woke us up at Neumann to think about these IT things I’m describing and for us to increase our software skills. Neumann hasn’t survived and thrived over 90 years by coincidence or luck. When there is good competition out there, we gotta be better or we’re not worth the money we are asking. Up to now, we think the success tells us that we are not doing the wrong thing. AT: As you ramp up your software and digital capabilities, how does Neumann consider its Solution-D foray from the early 2000s? WF: I understand why you are asking this question with a smile on your face. I can smile about it myself. There were years when I didn’t think there was anything funny about it. And the key word is AES42. This standard has not made it into the market, neither for the console manufacturers nor for any other outboard gear. And having not accepted the AES42 standard it was quite hard to offer an AES42-based digital microphones to a market which does not exist — at least not as a mass market. At the same time, Neumann has offered a variety of digital microphone interfaces. What’s more, Solution-D customers have been very happy with the product. When you go to the Philharmonie de Paris you will see Solution-D microphones in the ceiling, and they’re used week in, week out. AT: Sure. But apart from being philosophical about its fate, what were the technical Solution-D ‘take aways’? WF: We learnt a lot about signal processing, which we knew anyway from our times spent making mixing consoles. We know what a de-esser function is, we know how to make a good equaliser, and all the other features of a normal channel in a good mixing console. We still have the technical drafts and we will use that knowledge in future but not in an AES42-based microphone. Which doesn’t mean we won’t offer digital microphones in the future… maybe according to new standards. The other important learnings from Solution-D was the conversion process to turn an analogue signal into a digital signal, not somewhere in the signal processing chain but right behind the microphone capsule, inside the microphone. It meant Solution-D was able to offer a lot of mixing console-type functions — EQ and de-essing, for example — as part of the electronic circuit built

into the microphone. So the microphone had a connection to a software channel which you could see on your laptop and you could, for example, control the gain setting from zero to 63dB in one dB steps or you could change the polar pattern between 15 different patterns with a pushbutton. Those things did not have any influence on the self-noise level of the microphone because there is no self noise — once it is zeros and ones, the signal remains the same. As long as it is an analogue signal you always increase the self noise level by the time you put the gain up. So those things are predictable developments even in the future, when you talk about digital microphone technology be it Solution-D or something else. AT: You’ve been with Neumann since the early ’90s, how has your constituency changed? WF: The magic term here is ‘customer loyalty’; now in a group of younger customers. A few years ago audio engineers were still coming up through professional recording studios and they were all in contact with Neumann products and they knew that without Neumann the results wouldn’t be the same. Increasingly, people are bypassing the professional studio world. They’re still achieving more or less good quality recordings but they are not in contact with Neumann so much anymore. That is something we do sense and that is something we do see. That’s why we’re increasing our social media activities dramatically right now. As we get online, people who are not even audio engineers are making serious requests for our products and services. We have to stay on top of these developments. We can’t afford to think: the past was easier. Neumann needs these fresh thoughts, fresh ways of thinking and fresh young people who are users of multimedia. AT: Tricky waters to negotiate. A blogger doesn’t want to spend more than a couple of hundred bucks on a USB mic. WF: When we are targeting these younger target groups where we say ‘okay we need to have services and hardware solutions and software solutions for their needs of recording’, we are in a different field of business and that is something which I will not explain in detail because that is exactly our homework right now. But it means it’s a big change for every manufacturer and the earlier we face it, the better. If we don’t change, we lose business. AT: Readers of this magazine could point to examples of companies with great brands, going downmarket unsuccessfully. No one wants to see a Chinese-made pair of cheap headphones with Marshall’s distinctive logo on them, for example. WF: True. Interesting you use that example. Because there’s nothing wrong with thinking Neumann could make a headphone. We know how to make good transducers and the Sennheiser mothership makes headphones. I think people would consider such a product to be serious. The question is: would it be a 49 Euro set for a cell phone or would it be a real headphone. The answer is: it will be a real headphone and you can see it at NAMM 2019. AT: Thanks for the scoop Wolfgang! See you at NAMM.


BRING YOUR CREATIVE PROJECT TO LIFE EARN YOUR MASTER OF CREATIVE INDUSTRIES ALONG THE WAY FIND OUT MORE jmc.academy/masteraudio

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FEATURE

From Migos, to N.E.R.D., to Cardi B, to Beyoncé, Leslie Braithwaite is all over the sound of modern hip hop and R&B. His secret? He doesn’t fight to get out of the rough. Story: Paul Tingen

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While many top end mixers have a preworkout routine that can include a lot of prep work; like conforming a session to a preferred template, replacing existing plug-ins with preferred options or stripping them out to the bare track, Leslie Braithwaite reckons it’s all non-essential for getting the mix right and working fast. What’s important? Giving the artist what they want, which can usually be found in the rough mix. “Whenever someone sends me a session, the first thing I want to know is: Has the client been listening to the rough?” explained Braithwaite. “If they’ve been listening to it for a long time, I’ll put it on a track in the session, so I can really scrutinise it and lock into what they love about this rough. Then I can A/B it with what I’m doing. Someone who has been listening to a rough for a long time is not going to want something radically different. Especially in my world — the hip-hop, R&B and pop world, where I work with artists like Cardi B and Beyoncé — this is really important. “With Cardi B, Beyoncé or Ariana Grande, I will work from where they left off. They don’t like things to sound too different from what they had. I will often stick to the existing plug-ins, even where I’d tend to use something different, because in this genre, mixing is more about making it work and getting the vibe right than about getting things to sound perfect. I also don’t typically spend a lot of time trying to retrofit a session I have received to how I like things. I don’t adapt the session to my own template, or start the mix from scratch. I’LL WORK WITHIN HOW THEY’VE ORGANISED SESSIONS, ONLY OCCASIONALLY ALTERING ELEMENTS SLIGHTLY TO FIT WHAT I AM USED TO. THAT MEANS I CAN WORK A LOT QUICKER.

“For instance, if I open up a Cardi B session and she has four lead vocals bussed to one aux track, and that aux track is bussed to something else, I’ll just follow the flow of what they’ve done and see if

ABOUT BRATHWAITE Brathwaite’s current minimalist setup and approach is the end result of several decades of adapting to new technology, and whittling things down to essentials. Originally from the Virgin Islands, he attended the Recording Arts program at Full Sail in Florida and graduated in 1992. Following this he moved to Atlanta, where he worked with the likes of Jermaine Dupri, Babyface, LA Reid, and Dallas Austin, in the latter’s DARP studio, where analogue tape and big desks still ruled. Brathwaite moved to Patchwork studios in Atlanta in 2001 and to Akon’s Music Box facility in 2012. His credits over the years also include the likes of Aretha Franklin, Björk, and Eminem, and recent achievements include mixing N.E.R.D’s No One Ever Really Dies, almost the whole of Cardi B’s Invasion of Privacy, several songs on The Carters’ Everything Is Love, and seven tracks on Ariana Grande’s forthcoming album, some of them comixed with Phil Tan.

it makes sense to me. If I decide to use a different de-esser than the one in the session, I’ll make sure it sounds similar. Or if there’s a DVerb, I may replace that with a similar sound from my favourite vocal reverb, Altiverb, but one that’s a little richer and fuller to my ears. It does help that I’m used to the way certain engineers — like Evan LaRay with Cardi B, and Mike Larson with Pharrell Williams — organise their sessions. “It’s different if I’m working with Pharrell because he rarely gets attached to the rough. Often when I get a session from him, he’ll have finished it the night before! It means I have more freedom during the mix. Pharrell and Mike also normally send me sessions without many plug-ins, because they haven’t worked on it long, and that also gives me a lot of creative space. “The standard plug-ins you get in sessions today are by Waves, UAD, SoundToys, iZotope, Focusrite, SPL and so on. Everybody has them. But if I receive a session with plug-ins I don’t have, I tend to buy them real quick. It often teaches me new things, and it’s how I discovered the Turnado plug-in by Sugar Bytes. IT’S A COOL LEARNING MECHANISM, BECAUSE IT GIVES ME THE OPPORTUNITY TO HEAR WHAT PLUG-INS NEW, YOUNGER ENGINEERS ARE PICKING.”

LIGHT UP ’N’ EASY

Brathwaite’s no-nonsense approach is carried through into his stripped-back Atlanta mixing space. It’s in a building owned by Akon, who has an adjacent studio. However, with Akon spending a lot of his time on humanitarian efforts — like partnering with Shell to bring electricity and running water to a whopping 600 million Africans — Brathwaite tends to have the space, which has a recording room attached, entirely to himself. His futuristic studio looks like it was dreamt up by a Hollywood sci-fi special effects department. The few pieces of gear are simply the Slate Raven MTi2 touchscreen, Universal Audio Apollo 8 interface, Antelope Isochrone OCX Audio Master Clock, and Focal Twin6 B monitors with a KRK sub. “I used to have Yamaha NS10s, but don’t use them anymore, and my white KRK Rockets also don’t see much action,” said Braithwaite. “I fell in love with the Focals six years ago, and never looked back. I used to have a Mac tower, but I now run everything off the iMac. Being old school, and having worked for many years on an SSL, I like to touch things, but I don’t actually use the Raven touchscreen function much, apart from some convenience functions like batch commands, which assign a whole lot of commands to one button. I moved fully in the box about six or seven years ago, because doing recalls on a large format console became a pain, and working in the box is faster, and also better for smaller budget projects. My flow is so easy now that I don’t need an assistant anymore. Also, the UAD plug-ins give me the sonic warmth and depth I’m after. “Mostly, going in the box was about being able to work faster and being more productive. If you eliminate organising, cleaning up mixes and printing them, the core of mixing really does not take that long. Perhaps an hour. IF YOU SPEND MORE

In this genre, mixing is more about making it work and getting the vibe right than about getting things to sound perfect

THAN A COUPLE OF CORE HOURS MIXING A SONG, YOU CAN UNRAVEL A MIX. It really does not take that long to get

the sounds you really need. Mixing in the box has made many things faster, but what still takes a lot of time is getting feedback from artists and producers, particularly if they change their minds, and want to add new tracks or replace things.”

HAVING THE 808s NUMBER

The less is more approach also extends to the music Brathwaite primarily works with. The Atlanta trap variation of hip hop — with its dominant 808 bass and hi-hats, often no actual bass-line, and very little musical content and mid-range — strongly influences R&B and pop as well. Based in Atlanta, Brathwaite is right in the middle of this development, and he explains that balancing the kick against the 808 has become the new low-end challenge, while the lack of mid-range instruments tends to make mixing in the vocals much easier. Despite it being easier to find space for the vocal than in the past, Brathwaite still holds on to one of his own tried and tested working methods, which puts him at odds with many of his colleagues who tend to start their mix with the kick. “During my entire mix process, I am mentally mixing everything around the vocal,” he explained. “When I start work on a session, I will first mute all the instruments and start with the vocals. As opposed to getting a good mix of the music and fitting the vocals in with that, I think in the opposite way. Everything I do is based on getting the vocals to sound amazing and fitting everything else around that. When I hear a track the first time, I already know with a certain level of certainty how every part of the arrangement will affect everything else. So I’m always thinking about all aspects of the session, and where I need to compensate and where I don’t. Every stage of mixing has a thought process behind it, and you’re also thinking about all the other thought processes to occur after that. “So when I start a session and solo the vocals, I’ll first clean them up and make sure they are tuned as required. I often work with very experienced engineers, but they don’t always have the time to clean up every single thing. For example, with the Cardi B album, Evan was recording as fast as I was mixing. Often the vocals I receive are tuned, but based on the relationship I have with the client I know whether to tune them some more and mess AT 27


It’s all about the vocals for Braithwaite’s clients. Braithwaite: “I may put on a Waves Renaissance compressor, with a 3:1 ratio, and back it off a little bit. After that I’ll have the Waves SSL EQ, or sometimes the UAD SSL. Most microphones represent the low end in a weird way, so I usually cut some low end. Then I work on the high-mid frequencies, because some of them can be annoying. Then I’ll bus the vocals to an aux, and that aux will usually have a second de-esser, working lightly, and then another compressor, maybe the UAD Fairchild 670 Legacy set to default to add a little colour and presence.”

with them a little bit, or leave that aspect alone. In any case, tuned vocals are part of today’s sound. ARTISTS LIKE MIGOS WANT THE ANTARES AUTOTUNE ROBOTIC SOUND, BUT EVEN A GREAT SINGER LIKE ARIANA WANTED A LITTLE TUNING ON HER VOCALS. When we were mixing Beyoncé, she also wanted some tuning, not to correct her vocals, but because she wanted the sound and feel of AutoTune. “When people want that robotic sound, I use AutoTune, but when I’m dealing with someone who sings really well but there are a couple of bad notes, I’ll reach for Melodyne. I don’t use Melodyne as a plug-in, instead I use the standalone version, and then reimport the tuned vocals into the session. In my experience, the real-time tuning process is a bit hit and miss. During some passes it may miss a tuning issue, or it doesn’t tune the note the same every time you play it. It may even have weird glitches on it now and then, so I always print tuned vocals in the session. I do the same for all effects that are random to some degree, like flanging from the MetaFlanger or an autopan plugin. You don’t want a word on the right during one playback and on the left during the next.” AT 28

HOW TO KEEP THE VIBE Brathwaite makes a big claim when he says “mixing is about making it work and getting the vibe right, more than about getting things to sound perfect.” He elaborated on his point, saying “vibe is all-important these days. If you think about what someone like Drake does, there’s no structural intensity in terms of dynamics or buildups or breakdowns. It’s totally different. It’s more about the vibe. The vibe is usually already there in the material I receive, so let me let you in on a little secret. This is what I usually do to always stay interested and emotionally fulfilled by the music: I mix what the clients want and what they want to hear, and if I have the time I will do a personal mix. I do another mix where I will play with some

things, and those mixes I keep for myself… the rest of the world will never hear them. These alternate mixes help me to remain emotionally locked in and happy, and feeling like I am playing with new toys and learning stuff. It can get a little mundane if you are always sticking to the vibe the clients want, especially as it often does not require a lot. I do personal mixes for myself. Sometimes I do drastic things, like putting on all kinds of sweetening effects, or chop things up, or do different breakdowns, just to play around and have some fun. I will only play these mixes to the client if I feel they’ll be receptive. Sometimes people will ask me to try something, but most of the time it is for my personal satisfaction.”


AT 29


Leslie Braithwaite at work in his Music Box Studios space.

VOCAL ON DOWN

Once the vocals are cleaned up and tuned, Braithwaite engages some vocal treatments before moving on to other parts of the mix. “My vocal chain starts with a de-esser, usually the Waves Renaissance, and I then add a little bit of compression,” he said. “I tend to fix level issues with vocal rides, but I may put on a Waves Renaissance compressor, with a 3:1 ratio, and back it off a little bit. After that I’ll have the Waves SSL EQ, or sometimes the UAD SSL. MOST MICROPHONES REPRESENT THE LOW END IN A WEIRD WAY, SO I USUALLY CUT SOME LOW END. Then I work on the high-mid frequencies, because some of them can be annoying. “Then I’ll bus the vocals to an aux, and that aux will usually have a second de-esser, working lightly, and then another compressor, maybe the UAD Fairchild 670 Legacy set to default to add a little colour and presence. Then, if I want to add some sheen after the de-essing, I’ll engage the UAD Precision EQ and boost just 1dB at 27K. This pulls everything up in the higher regions to give a nice, smooth high end. “My vocal reverb of choice is the Altiverb. I love the smoothness of it, and with so many different impulse responses you can find all kinds of neat things. I’m a visual person, and I like the aesthetic and being able to see what the room looks like. My vocal delay will be the Waves HDelay. Ping-pong delays are not my preference, but if there’s one in the session I’ll honour it. “Once I’ve got the vocals the way I want them, I’ll get some general vocal levels, but unless a word really pops out, I won’t do any involved vocal rides, because they depend on whatever else happens around the vocal. My next step is to work on the AT 30

kick and 808. In a lot of hip hop and pop music these days the 808 acts like the bass, so THE FIGHT YOU HAD BACK IN THE DAY OF THE BASS AND KICK HAS NOW BECOME THE FIGHT BETWEEN THE 808 AND THE KICK. The two usually work well together, because they have different properties and are in different frequency ranges. In Migos’ and a lot of Cardi B’s music, the 808 and the hi-hats drive the song together. “My plug-in choices for the 808 usually start with the UAD Little Labs Voice of God bass resonance tool. I don’t often tweak any of the other parameters besides the amplitude. The other thing I might do is run the 808 through a UAD Pultec EQ or the Waves Puigtec EQ, just to add some 30Hz or 60Hz for some more warmth, depending on the 808, but not too much. I usually boost the kick between 60Hz and 100Hz, using the UAD Pultec EQP-1A. Those are my weapons of choice on the 808 and kick. “In general I don’t add kick or snare samples, nor do I use a lot of plug-ins. Many artists are so tuned into the rough, they don’t want the sounds dramatically changed. So putting on a whole load of plug-ins, or adding samples, can be counterproductive. After that I’ll get to the bass, if there is one. I’ll probably also run it through a UAD Fairchild 670 Legacy. The bass preset on that works really well for many bass sounds. So I’ll get the low end of kick, 808 and/or bass working right, and then I’ll bring in the snare and the hi-hats. I don’t do much to them, sometimes I’ll add some high end to get some more crack. “From there I’ll bring in the other instruments. In the case of Cardi B’s Money, the main driving instrument is the piano, so I have to make sure that rings through. In Money, all I did was add some high end to the piano, to make sure it cut through a bit more. In a lot of R&B, hip hop and

pop these days there are only one or two driving instruments, which gives a lot of space to the vocals. Once I have the drums and the music sounding the way I want it, I bring the vocals back in, and do the final vocal rides.” CLEAN MASTER

The final mix stage involves Brathwaite getting feedback on his mixes, and then printing them and sending them to the mastering engineer. Once again, he has a few approaches that are well out of the mainstream, most notably not putting anything on the master bus. “ONE OF SEVERAL THINGS I DO VERY EARLY ON IN THE MIX PROCESS IS FIGURE OUT WHO IS IN CHARGE, WHO AM I ANSWERING TO, WHO WILL DO THE FINAL APPROVAL ON A MIX. This varies from situation to situation. With Pharrell, it’s him, but if it’s Pharrell and Beyoncé, like on the Everything Is Love album, it’s Beyoncé. With Cardi B, it is Craig Kallman, head of Atlantic. Cardi also gives feedback too and approves mixes, but Craig has the final say. “When I send my mixes out to my clients, I may put the Slate FG-X mastering process on the master bus, just to boost volume, but when I’m printing my mixes for the mastering engineer, I don’t put anything on. I don’t want to get involved in the whole loudness issue, and I prefer to give the mastering engineer the space to do their thing. I know that many mixers feel pushed to come in at the level of the rough, or louder, but because you are in control of the technical environment, you have to guide the artist. I tell them: ‘Don’t judge this mix as it is, wait until it has been mastered, and then judge.’ I feel that mixing and mastering are two very different disciplines, and I like to stick to the mixing component!”


AT 31


REPORT

Product Launch Report For the first time, dBTechnologies’ new flagship line array left Italy for a special APAC demo on the shores of a Bali beach. Review: Preshan John

Budget plastic powered loudspeakers have been good to dBTechnologies. Scratch that, they’ve been great! But for the last few years, the Italian manufacturer has been turning that brand image on its head. Sure, its bread ’n’ butter Opera and BH active loudspeakers keep selling like hotcakes, but more premium-spec’d products like DVA and Vio have been winning over the mid-range rental and installation markets. With a near-irresistible value proposition, it’s not hard to see why. It’s not marketing fluff either; dBTechnologies’ sales in the APAC region have grown an astonishing 12-fold in the past five years. To highlight this success, and to join some dots between products and faces, dBTechnologies AT 32

organised its first-ever APAC Distributor Meeting in Bali this September. It brought together representatives from several countries who flew in for the three-day event held at a resort in Badung. dBTechnologies also flew in staff from all of its key departments. It was the perfect opportunity for local distributors to get to know the faces behind the emails. More importantly, it was a chance for dBTechnologies to instil its aspirational vision into the hearts of its partners. VIO L212

Key to that vision is the new jewel in the dBTechnologies crown — Vio L212. It’s the company’s first full-scale line array module, following on from the release of its little siblings, Vio L210 and L208, over the past year. The

Bali launch was the first time the L212 had left Italian shores and the staff were gushing with enthusiasm for their flagship product. APAC Sales Manager Manuele Poli calls it “a game changer for dBTechnologies.” Vio L212 is a three-way box comprised of dual 12-inch woofers with three-inch voice coils, four 6.5-inch midrange woofers with two-inch voice coils, and two 1.4-inch neodymium compression drivers with three-inch voice coils. All those transducers are packed tightly together for maximum phase coherency. As Product Specialist Marco Cantalú explains, “Having all of the sources so close to each other, we could really make a great improvement on the off axis reproduction. It’s a condensed collection of technology in the most compact and lightweight size we could conceive.”


based on a three-point system. Two front links connect the modules together while the central rear strand lets you pre-select splay angles between 0.58° using a hook-type link. A number of accessory options let you transport, rig and protect the system in whatever way suits you best. AIRING IT OUT

ALL IN THE BOX

A primary point of difference with dBTechnologies line arrays is they’re all powered. Each Vio L212 is equipped with two Class D Digipro G4 amplifiers to provide a total of 3200W RMS per box. The switched mode power supply is equipped with PFC (Power Factor Corrector) to improve the system’s efficiency, and the L212 accepts audio via Dante. The use of NFC is a neat innovation. The technology (typically used for mobile payments or file-sharing between phones) has been implemented in L212 with proximity sensors that determine the position of each box in an array. Pair this with the remote control Aurora Net software and you can easily identify and match each box’s physical position — sure to be handy when touring. Rigging is the same as smaller Vio systems and

As far as PA demonstration conditions go, the Bali launch of the Vio L212 was near on perfection. Rear reflections weren’t an issue here — the two L212 hangs of 12-a-side with 18 S218 groundstacked subwoofers fired out over 100m of unobstructed lawns, then straight into the ocean. Visitors could roam around weighing up the system’s throw and coverage. A variety of program playback was followed by a local cover band who rocked out for the rest of the evening. Hearing the rig with a live band gave a sense of the system’s ability to project a wide dynamic range with presence and impact. The L212’s tone feels pleasantly uncoloured and it didn’t grow fatiguing. Even more impressive is the pristine preservation of high end at distances over 80m. The S218 subs are clean and tight with excellent rear cancellation in cardioid configuration. THE BIG PICTURE

So what’s the end game for this ambitious company? Manuele emphatically states that dBTechnologies has no interest in competing with the PA giants. Instead, the vision is to hit the mid-range pro audio market hard with the intent of becoming the leading PA supplier in that market segment. dBTechnologies hopes that by achieving that goal its products will become attractive alternatives for the lower end of the top-range market. In other words, if you run a rental company with an inventory that’ll handle the next Foo Fighters show, chances are you’re not on the waiting list for a Vio demonstration. Although you may fancy a Vio rig as a backup offering when all your L-Acoustics and d&b stock is on the road.

S218 SUB dBTechnologies released the new S218 dual 18-inch neodymium subwoofer alongside the L212 announcement as the perfect partnership of tops and lows. The S218 response descends down to 28Hz and features much of the same innovations as the L212 such as a switched mode power supply with PFC, Dante compatibility, and NFC proximity sensors to inform Aurora Net of each sub’s position. Onboard DSP includes a cardioid stack preset button which automatically processes the sound of the backward sub in gradient inverted stack configurations to reach maximum cancellation on the rear side.

On the other hand if your rental company is looking to step up from supplying birthday parties and smaller corporate gigs, dBTechnologies wants its line array systems to be positioned as your premium option. Vio systems are priced very competitively. After spending three days with dBTechnologies distributors, a common catch phrase seemed to be “a very high performance-to-price ratio.” You get a lot of PA for what you pay. And that’s enough to put heat on the big players and how they pitch their lower level offerings. “We really brought the best engineers in firmware, software, acoustic design, transducer design and mechanical design,” says Marco. “Together these brought the realisation of the Ferrari of the line array system.” That is, loud, proud, stylish and top of the line from the Italian manufacturer. Exclusive, just not in the rarified air of a Bugatti. www.dbtechnologies.com www.nas.solutions AT 33


TUTORIAL

VIEW FROM THE BENCH

Is Class A the best, or merely the first? Column: Andy Szikla

+VCC DC Current AMPLIFIED AUDIO

LOAD RESISTOR

Vout

ELECTROMAGNET PLUNGER

WEAK AUDIO

CARBON ELEMENT

Vin

0v

0v

DIAGRAM

I have consumed too much peyote and am going to die. In my dream haze I see visions of the jungle with its cherry red sky and lemon sun blazing through every fringe. I push a branch aside and step into a clearing where a crowd of people are mingling, shaking hands, and exchanging business cards. Wiping away sweat, I take a heavy box from under my arm and place it carefully on the table in front of me. Inside this box is the best amplifier ever made in the history of everything. It was given to me by an electric serpent at the beginning of time, and I have carried it many miles. Mrs Tech Bench appears at my table and tells me she is there to receive the amplifier. She says she has seen it in her dreams, countless times. Her outstretched hand tempts me with a bushel of cash, but when I reach for it she draws away suspiciously, looks into my eyes, and queries: “Is it Class A?” “What do you care if it’s Class A?” “Class A is the best”, she says. I probe further, “Do you even know what Class A means?” She stares at me like I am an idiot. Spurred, I clamber onto my high horse and rant with rising intensity, “Class A is not a badge of achievement. The good amplifiers do not begin AT 34

CARBON CLASS A

at Class A and then descend in quality towards Class Z. Amplifiers are classified in terms of their architecture and operating characteristics, and in that context the term ‘amplifier class’ is interchangeable with ‘amplifier type’. The quality of listening experience one can expect owes much less to the operational class than to the specific design, and it is possible to create good and bad designs in any class. Did you know that some of the crappiest musical greeting cards use Class A amplifiers, and many elite powered studio monitors use Class D. A designer will select an appropriate class based on what their amplifier is trying to achieve...” By the time I draw breath, Mrs Tech Bench has transformed into Aphasia, the wombat spirit, and burrowed deep into the jungle floor where she can no longer hear my racket. I’ve lost the sale. CLASS A

So what is Class A, if not the zenith of all classes? Well, it is merely the oldest. Before transistors there were valves, and before valves there were mechanical carbon amplifiers — connected in Class A. Diagram shows a scheme for telephone line repeaters, used by Bell Telephone as far back as 1901. In a Class A amplifier the output transistor

(or valve, or carbon element) is turned on all the time, even when no signal is present. This allows an audio wiggle at the input to enjoy continuous attention as the internal electronics follow along without interruption. Pretty much all other amplifier classes chop signals into bits that get processed separately and then re-assembled, which invites glitches. When dealing with very small signals like those from a microphone, it stands to reason that chopping one to bits and then making it 1000 times bigger might risk amplifying the glitches as well. That’s not a problem for Class A, since the signal is processed whole. This is considered its principal advantage. In fact, I don’t believe I’ve come across any small-signal audio input that was not configured in Class A. However, the acts of driving line outputs and speaker loads are now dominated by more recent audio classes. THE CONCEIT WITH CLASS A IS THAT SINCE THE SIGNAL IS NOT INTERRUPTED, THE OUTPUT SHOULD BE A LINEAR RECREATION OF THE INPUT, AND IT GENERALLY IS, MORE OR LESS. In fact, Class A’s harmonic distortion performance can be hard to beat. The concept of ‘linearity’ is worth explaining. If say the signal at your amplifier’s input doubles in amplitude, then the output does likewise. Given


+VCC

AMPLIFIED AUDIO

LOAD RESISTOR (Pulls Up)

Vout WEAK AUDIO TRANSISTOR (Pulls Down)

Vin

0v

0v

DIAGRAM

CLASS A

a further 20% increase, and the output follows exactly. This behaviour is said to be ‘linear’ because the input and output signals wind up the same shape, with one simply bigger than the other. Now imagine if the input amplitude doubles, but the output only increases by 97%. Then it doubles again but this time the output only increases by 70%. This uneven transfer of signal trace from input to output is said to be ‘non-linear’. It is important to understand that in the above examples we are not just talking about when the music gets louder or softer; we are talking about the rise and fall of every cycle within the audio wave, and the resultant mismatch is called ‘harmonic distortion’. Transistors, valves and carbon elements exhibit good linearity only within a portion of their overall operational range, and beyond that, amplification can be a little wobbly or compressed; sometimes on purpose. In the Fairchild 670 compressor, gain reduction is produced by moving the valves into electronic non-linearity. Circuit architecture itself can offer challenges. IN ITS PUREST FORM, CLASS A IS INHERENTLY LOPSIDED, WITH OUTPUT PEAKS PULLED IN ONE DIRECTION BY A POWERFUL TRANSISTOR OR VALVE, AND IN THE OTHER DIRECTION BY A PASSIVE RESISTOR AT

When things get extreme, the result can be a lopsided signal like in this oscilloscope shot of a ‘corrupted-for-thepurpose’ Class A transistor amp . The signals in and out are overlaid on each other to show their difference, and the mismatch corresponds to about 10% harmonic distortion. In some old-time Class A valve amplifiers, asymmetrical harmonic distortion is more common than you might think, and since curvy distortion contains lots of even harmonics, it actually doesn’t sound that bad. In fact, it may be the thing we love best about some of that equipment. Other classes tend to distort more edgy and square, which can sound less pleasant, unless you’re a fan of German metal. Class A has the advantage of only needing a single power supply, but the main drawback is massive inefficiency. Because the active devices are always conducting, and never turn off, most energy winds up dissipated as heat and only a small part drives the signal — typically around 30% or so. In an effort to achieve more efficiency, less heat, and better symmetry, old-time designers went back to their drawing-boards. One idea was to connect two Class A amplifiers in an opposing ‘push-pull’ arrangement, which we today refer to as ‘bridge mode’. Another was to use an active device called a ‘current-source’ in place of the load resistor. Both these techniques deliver superior harmonic fidelity, with the push-pull idea providing near perfect symmetry. THE MERCY OF DOWNSTREAM LOADS

CH1

INPUT

CH1

OUTPUT

1000.00 Hz

DIAGRAM

+VCC

AUDIO IN

AUDIO OUT

TRANSISTOR (Pulls Up)

Vin

Vout

CLASS B TRANSISTOR (Pulls Down)

Crossover Distortion

–VEE

DIAGRAM

CLASS B

The logical next step was to design push-pull symmetry into a single amplifier. Initially appearing around 1930 (using valves), Class B dispenses with the load resistor, and employs two opposing transistors at its output — one pulls down and the other pulls up (maybe they should be called ‘pull-pull’ instead) . Each transistor only conducts for half the signal cycle, and both devices are turned off when there AT 35


+VCC

Class A is not a badge of achievement. The good amplifiers do not begin at Class A and then descend in quality towards Class Z

AUDIO OUT

TRANSISTOR (Pulls Up)

+

AUDIO IN

@1.2 Volts

Vout

− TRANSISTOR (Pulls Down)

Vin

is none, so the result is greater efficiency (around 50%) and less heat. The major drawback with Class B is that every time the signal crosses zero, both transistors are completely off, so part of the signal goes missing. This is called ‘crossover distortion’, and it has nothing to do with speaker crossovers. IT

No Crossover Distortion

–VEE

DIAGRAM

CLASS AB

TAKES THE FORM OF PERSISTENT SQUARE CLIPPING RIGHT IN THE MIDDLE OF THE SIGNAL (RATHER THAN AT THE PEAKS) AND IT MAKES CLASS B A VERY LO-FI ANIMAL INDEED.

+VCC

CLASS AB

The solution to Class B crossover distortion is to doctor the circuit slightly so that instead of both transistors being turned off during zero-crossings, they are both turned on. That means one device handles the upper half of the wave, both devices work together across the middle, and the other device handles the lower half of the wave. Kind of like a baton being passed in a relay. Thus none of the signal is dropped, and extremely low harmonic distortion figures are possible. SINCE BOTH

TUNED CIRCUIT

AMPLIFIED RF

Vout WEAK RF

TRANSISTORS ARE NOW ENGAGED FOR SLIGHTLY LONGER

TRANSISTOR

Vin

THAN HALF OF EACH CYCLE, THEY ARE SAID TO BE BIASED TOWARDS CLASS A, SO THIS TOPOLOGY GETS REFERRED TO AS CLASS AB.

The low output impedance and excellent symmetry of Class AB makes it perfect for driving speakers, transformers and line outputs, so eventually it turned up in everything, and is arguably the most ubiquitous amplifier architecture of the last 50 years. Go back to say 1985, and almost every hi-fi amp, professional sound reinforcement amp, mixing desk, radio, telephone, and audio op-amp was configured in Class AB. Even the Urei 1176 Compressor, revered for its Class A-ness, uses a Class AB stage to drive the output transformer. On the downside, both Class B and AB normally require a second power supply rail, which adds some extra cost and clutter. CLASS C

Class C amplifiers operate at radio frequencies, and are not used for audio at all. The architecture looks a bit like Class A, but with the output resistor swapped for a tuned circuit usually made up of a capacitor and an inductor . Class C will amplify a specific frequency while attenuating all others, so it is commonly found in radio transmitters and receivers, and also forms the basis of some oscillator circuits used for logic clocks and digital timing. AT 36

0v

0v

DIAGRAM

CLASS D

The 50% practical efficiency of Class AB seemed like plenty, until the advent of cell phones and mobile music playback systems in the ’80s and ’90s. Batteries cost money, and rechargeables drain quickly (even now) so designers inevitably shifted their focus to power conservation, with the goal of 100% efficiency as their target — meaning every electron consumed by an amplifier would be used to drive headphones or speakers, with zero left over to dissipate as heat. Looked at another way, if efficiency increases from 50% to 100% then your battery will last twice as long, and everyone with a smart phone wants that. CLASS D (FIRST MOOTED IN 1958) TODAY BOASTS EFFICIENCIES OF WELL OVER 90% AND HAS REVOLUTIONISED AUDIO AMPLIFIER DESIGN. Sorry,

CLASS C

the ‘D’ does not stand for ‘Digital’ — Class D does its business in analogue, but goes about it in a nonlinear way, a bit like a switch-mode power supply. Class D uses output switching to rapidly turn transistors (usually MOSFETs) completely on and off. This switching happens at a rate much higher than audio frequencies, so that each switching cycle (called a ‘duty cycle’) handles a very small section of the signal waveform. At a rate of say 200kHz, a 20kHz sine wave would be split into 10 equal sections, and a 20Hz sine wave into 10,000. When switched on, the transistors cause the output voltage to rise by supplying current (via a low-pass filter) to a loudspeaker or other load. A comparison circuit monitors this rise, and when the output gets to the right level (microseconds later) the transistor


AT 37


turns off and waits for the next switching cycle where the process repeats. In my diagram, the input signal is compared to a continuous sawtooth wave whose frequency sets the duty cycle rate . Because the ‘pulse’ duration within each duty cycle varies according to demand, this kind of energy delivery is known as ‘pulse-width modulation’. In an alternative variant of Class D the pulse duration is fixed, but the length of the duty cycle varies, and this scheme is called ‘pulse-density modulation’. What both schemes have in common is that they spit tiny pellets of energy at a load, to build mountains of amplitude – a bit like a 3D printer making an object out of droplets of plastic. Because the transistors are only ever turned on for the amount of time required to produce each droplet, there is very little wasted energy, and very little heat, meaning amplifiers can be made smaller because they don’t need a great big ventilation system. In the late nineties and early noughties, Class D schemes really came of age, making self-powered monitors practical, achievable, and affordable. In pro sound reinforcement these boxes were welcomed with open arms, so much so that it is now unusual to rig a live show without them. Also, Class D amplifiers are used in many current brands of near-field studio monitors, with good ones boasting less than 0.01% Total Harmonic Distortion. In smart phones and other headphonedriving, battery-powered gadgets, Class D is a popular inclusion for its low power consumption, small size, and cool running.

PULSES OUT

SAWTOOTH IN

AUDIO IN

AUDIO INPUT AUDIO OUTPUT SAWTOOTH GENERATOR

COMPARISON AMP

OUTPUT SWITCH

DIAGRAM

LOW PASS FILTER

CLASS D

+VCC POWER RAIL MODULATION +High Rail TRANSISTOR (Pulls Up)

AUDIO IN

+Low Rail

+ @1.2 Volts

CLASSES E & F

+VCC

+VCC

Vout

AUDIO OUT

0

Now we’re back to radio frequencies. Class E is like a mash-up between the tuned linear circuit of Class C and the full on/off switching of Class D. They are designed to be used above 100MHz, where Class C starts to struggle. Class F is like E with fries.

TRANSISTOR (Pulls Down)

Vin

−Low Rail −High Rail

POWER RAIL MODULATION

–VEE

–VEE

CLASS G (Power Rail Switching)

CLASS H (Power Rail Tracking)

–VEE

CLASS G

This variant of Class AB was borrowed from some hi-fi amps like the NAD 3220 from 1989, and used to great effect in mobile phones. The basic principle is to use two separate sets of power supply rails — some lower voltage ones for the vast bulk of audio demands, and higher ones rigged to suddenly switch on only for louder peaks and head-banging. Since the higher voltage rails are completely off most of the time, the dividend is lower power consumption, less heat, and longer battery life . CLASS H

DIAGRAM

and there is no output — but when the pulse-width on one is wider than the other, the output will swing towards it. Crown say this is more efficient than regular push-pull versions of Class D, and solves the problem of output transistors blowing up if they accidentally turn on at the same time. CLASS S

Used in some cell phones, this scheme takes the idea of Class G to the logical limit. Instead of rail switching, Class H tracks the input signal and moves a single set of power rails up and down in unison. This leads to around 80% efficiency, which is better than Classes G and AB, but not quite as good as D.

Another version of Class D that is usually associated with radio frequency or digital input audio operation. It applies a fancy algorithm to convert multi-bit digital signals directly into pulse-width modulation, bypassing the normal D-to-A stage. Annoyingly out of sequence, the ‘S’ is generally believed to stand for ‘Switching’.

CLASS I

CLASS T

A push-pull variation of Class D, patented by Crown Amplifiers, using separate pulse-width modulation on two opposing switches. When the state of both switches match, they cancel each other

Yet another glorified version of Class D, in a proprietary design by a company called Tripath — hence Class ‘T’ — from 1996. Apple used them in the G4 and iMac. Their central claim was that

AT 38

CLASS G & H

a superior process was employed to control the pulse-width modulation, with feedback taken from the switching stage rather than the output, resulting in higher gain, and lower distortion. ALL CLASS

This writer is not aware of any other meaningful amplifier classes, so most of the alphabet is still up for grabs, if you have any brain-waves. Not sure about the Tripath idea of simply co-opting any letter you like. Will this lead to Meyer, M-Audio, Motorola and Marshall all in litigation over Class M? Hmmm...


AT 39


REGULARS

PC Audio AT Team

When manufacturers and developers don’t see eye to eye, your fellow users can be the glue between hardware and software.

Friendly Neighbourhood Helper

Topics: 254 Replies: 3,105 Total: 3,359

Column: Martin Walker

Whatever PC-based software or hardware you have, the first port of call for assistance should always be the manufacturer/developer’s website. Many host online forums where you can interact with other users, and hopefully a company representative will also pop in from time to time when an ‘official answer’ is required. Others simply have support links on their websites, where they post official FAQs regarding issues that may crop up for each product they have, as well as news about updates and so on. However, when you are combining hardware and software from different sources, and neither company can provide you with answers, sometimes help can come from unexpected quarters. Time for a shout-out to those enthusiastic third parties (i.e. other users like you and me), who devote so much time and energy to solving such problems and letting others share in their solutions. GLUING SURFACES TO DAWS

I was reminded of this recently while wondering if I could improve the integration between my hardware control surface and DAW. I’d bought myself a Novation 25SL MkII (a very handy semiweighted, two-octave keyboard offering aftertouch plus an array of user-defined sliders, buttons and rotary pots), which sits immediately above my PC keyboard for editing duties. Its Automap software functions initially seemed extremely handy, assigning these hardware controls to various software parameters in my chosen DAW (Reaper), such as channel faders, pan controls, mute & solo buttons, plus a set of basic transport functions. Unfortunately I couldn’t find a more sophisticated Reaper/Automap template at the Novation website, but as often happens, a Reaper user named ‘Padre PC’ had taken up the challenge and created a really detailed Control Surface plug-in for his own use (https://forum.cockos.com/showthread. php?t=42928), which I discovered via an Internet search for ‘automap+reaper’. This offered me six presets, each containing four pages of hardware-controlled mappings, including track names picked up from my project, real values (such as dB for the faders and % for pan), the AT 40

MIDI Editor, Main controls such as X/Y zooming and scrolling in the Reaper arrange page, markers, Track-Item Editor functions, Track/Envelope Editor functions, and Assignable controls. This free download even included a PDF manual. So there’s your first big tip: If you’ve got two bits of gear you want to get the most out of, simply try an Internet search containing both their names for third party solutions. If this fails, refine your search by paying a visit to any official or unofficial forums relating to your particular DAW, and use their internal search engines. It may take longer, but you might also nab other user-created gifts along the way. MIDI HARDWARE STORE

User-created glue can also be invaluable between MIDI-enabled hardware (including synthesisers, drum machines, samplers and effects) and your PC. For instance, a fine selection of user-created helpmates can be found among the pages of the CTRLR website (ctrlr.org). The Ctrlr utility can control any MIDI-enabled hardware, and you can use it to create ‘panels’ (custom user interfaces) for all these products, and then host them as standalone editors, or VST plug-ins in your favourite DAW. Ctrlr is an open source project, so coders can add patches or new features themselves. There are plenty of dedicated users who have already created a wide variety of panels, many of which are almost photographic reproductions of their hardware front panels, and often far easier to operate than the hardware equivalents. You also get the benefits of computer-based preset handling, typed entry of patch names and the like. Some even include waveform and wavetable editors where applicable! There are already many Ctrlr panels available for hardware from Ensoniq, Korg, Novation, Roland, Sequential/DSI, Waldorf and Yamaha, as well as for plenty of modern boutique and retro hardware items. The vast majority are free to download, and in my experience can be a godsend for both patch creation and bank organisation. As well as utilities, hanging out in forums will also keep you up to date with any firmware updates, which can sometimes add new features, as well as dealing with the odd bug.

USER REAKTIONS

Some software developers even curate user contributions, allowing their creators to upload them to an official website, which certainly makes them easier to find. One of the best in my experience is Native Instruments. Its Reaktor User Library (www.native-instruments.com/ en/reaktor-community/reaktor-user-library) currently contains a wonderful collection of nearly 6000 devices, quite a few of which in my opinion rival commercial offerings. Thankfully, you can sort them by ‘most downloaded’ or ‘user rating’ as well as chronologically, which should help greatly in tracking down those that are of particular interest to you. SHARING IS CARING

Whatever music hardware products you have, even in the absence of official user libraries it’s well worth searching out those smaller unofficial websites that specifically cater to them, because you’ll invariably find a group of enthusiasts with talents ranging from amateur to professional, all eager to help each other solve problems and move forward together. User creations may well vary considerably in both stability and sophistication, but you’re also likely to find solutions to such bugs from those creators or on the forums where they hang out. Even if the original user is unable for personal reasons to carry on developing a utility, the open nature of these developments often means other coders take over the challenge. As far as support goes, it tends to be swings and roundabouts. It’s unusual for user creations to get well-written and detailed PDF manuals, but on the other hand you’re far more likely to be able to talk directly to that creator, explain your particular problem, and sometimes even change the direction of the product as other people chip in and agree with your suggestions. It’s good to share!


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REVIEW

UNIVERSAL AUDIO APOLLO X8 Audio Interface

Universal Audio’s freshly minted Apollo X range hits the spot with ‘HEXA Core’ processing, superior conversion and surround monitoring.

NEED TO KNOW

Review: Greg Walker

PRICE Expect to pay Apollo x6: $3299 Apollo x8: $3999 Apollo x8P: $4999 Apollo x16: $5499

AT 42

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

PROS Upgraded DSP, converters & clocking SMPTE standard +24dBu operation Compatible with older Thunderboltequipped models Talkback mic built in 5.1 and 7.1 monitoring capability

CONS Thunderbolt cable not supplied Still uses older processing cores Not compatible with older PCs No plans for surround UAD-2 plugs

SUMMARY The Apollo line continues to flourish with the latest hardware upgrades improving sonic and DSP processing performance. New surround monitoring, +24dB operation and built-in talkback mic add even more pro-requested features to an already impressive product.


APOLLO & THUNDERBOLT 3 – ARE YOU COMPATIBLE? There are a few potential issues to watch out for when connecting devices via Thunderbolt 3. The first is to check all connecting cables have the Thunderbolt lightning icon. The actual connector type is known as USB-C but not all USB-C cables are Thunderbolt-enabled. Mac computers with TB1 and TB2 ports can run the Apollo X range using a TB2-to-TB3 adapter. When purchasing an adapter ignore the fact that the data direction arrow points the wrong way, the adapters transmit data just fine in both directions. Unfortunately, PC users will need a TB3equipped computer to run the new Apollos.

Universal Audio launched its first silver Apollo units in early 2012. What was at that time an interesting new competitor in the crowded field of audio interfaces has since become an industry-leading behemoth with tens of thousands of units seeing use every day in both home studios and professional facilities. Key to the original Apollo’s success was its utilisation of advanced hardware modelling UAD plug-in software via built-in SHARC processing cores. While other companies were working along similar lines, the depth and sonic authenticity of the UAD reverbs, compressors and EQs was a real step up from the majority of plug-ins available at the time and the word quickly spread throughout the audio world. 2015 saw the first major hardware revision with the launch of MkII blackface units featuring improved specs and a slightly tweaked front panel layout. At the same time plug-in processing was upgraded to UAD-2 and the Unison preamp modelling software (dormant at the time of the original release) started to really come into its own with realistic emulations of classic Neve, API and UA preamp models giving users diverse tracking options from the same 1U box. Fast-forward to 2018 and UA has upped the stakes again with the release of the Apollo X range. The Apollo x6 is aimed towards the more budgetconscious with two mic preamps and a slightly smaller I/O count. The x8 reviewed here features four preamps and eight independent analogue I/Os to cover most recording bases while the x8p packs eight preamps for larger sessions. The x16 ditches the preamps altogether, instead offering 16 channels of line inputs and outputs on DB-25 ports. In keeping with previous Apollo models, the 16-channel model is fitted with the company’s flagship converters that will out-perform the rest of the new range, including the x8. The Apollo X range ships with the Realtime Analogue Classics Plus plug-in bundle and a oneyear parts and labour warranty. THE MORE THINGS CHANGE…

First impressions of Apollo x8 are that things haven’t changed much at all, at least externally. The build quality is excellent and there’s a new backlit logo stage left but otherwise this unit’s front panel layout is pretty much unchanged from the previous MkII model. Very Dark grey is the colour of choice which is fine by me (the original silver colour was always an irritant in the otherwise exclusively black membership of my racks). The economical yet simple-to-use rotary controls and buttons are laid out as before. Ditto the informative and clearly

For those contemplating expanding an existing system, the new models will interface with older Apollo units equipped with the Thunderbolt 1, 2 or 3 option cards, UAD-2 PCIe Cards, Apollo Twin Thunderbolts and UAD-2 Satellites (Firewire, Thunderbolt and USB). The only incompatible units are USB-model Twins, while Firewire-model Apollos need to be upgraded with the optional thunderbolt card before joining in. While the new units are designed to work with all major DAWs and computers its worth checking the UA website for compatibility with your particular system.

legible central backlit screen. Round the back wordclock, S/PDIF and ADAT ports are as-youwere with the only real difference from previous iterations being the pair of upgraded Thunderbolt 3 ports. The bottom line here is that the original layout was very effective and UA hasn’t seen the need to rejig things front or back. In designing the new range, UA’s engineers instead focused their energies on a number of key areas ‘under the hood’ including upgrading and standardising DSP processing power across all units, redesigning the power supply and adding extra features such as selectable SMPTE standard +24dBu operation, 7.1 surround sound monitoring capability and the handy addition of a built in talk-back mic. SIX IS THE MAGIC NUMBER

The headline acts of the new Apollo X range are undoubtedly the six processors snuggled on the PCBs of all four new models. Using UAD plug-ins is a highly addictive audio experience so it’s only natural that users want more processing power to harness more plug-in instances. While the number of UAD cores has grown to six with these new units, it is worth noting that ‘HEXA Core’ simply refers to six processors, not an upgrade of the processors themselves leading to some confusion and griping online. The same SHARC chips are still utilised meaning there is no performance improvement within each processing chip in the Apollo X range. Users who love stacking plug-ins on the preamp inputs will still experience the familiar limitation of only being able to utilise a maximum of one processor per input channel so there is no sharing of the load between processors in input mode. While powerhungry users may have issues with these limitations not being addressed on the new machines, a 50% increase in UAD-2 processing power over the old Quad models is nothing to sneeze at. It is also worth bearing in mind that UA has a track record of unleashing powerful new software capabilities from its existing hardware, so my educated guess is that the Apollo X’s new architecture will likely allow down-the-line feature upgrades. LOAD ’EM UP

So what can you do with six UAD-2 processors? Well as I discovered, quite a lot. I tried a few experiments with loading plug-ins into the Console channel inserts and managed 12 Neve 1073 preamp/EQs before I maxed out the DSP headroom. The less power-hungry EMT140 reverb (a personal favourite) managed 30 instances

while I got a sore finger clicking up 44 instances of the Fairchild 670 Legacy compressor and then discovered I’d only used up 37% of available DSP! Suffice to say there’s considerable processing power on offer and if, like me, you’ve been carefully spreading out your UAD plug-in instances on a less powerful system you’ll be able to let your hair down a little (or a lot). PREACHING TO THE CONVERTED

The designers’ other main priority with the X range was upgrading the conversion and clocking. UA’s converters have always sounded good but the Apollo X series delivers improved specs across the range with upgraded ESS Sabre chips and a redesign of their associated analogue circuitry. UA is claiming the sound of these new converters is ‘mastering grade’ with the x8’s signal to noise ratio 3dB higher than the mkII Apollo 8 on the AD side, 6dB higher for DA and an impressive 8dB higher on the monitor outputs. The x8 also boasts impressive total harmonic distortion figures of -113 dB (0.00022% @ 23 dBu) on the line inputs and -119 dB (0.00011% @ -1 dBFS) on the line outputs. Improvements in headroom mean the x6, x8 and x8p boast 129dB DA dynamic range, while the x16 ups the ante to 133 dB. UA’s new dual-crystal clocking circuitry offers improved stability by using separate clocks for 44.1kHz and 48kHz operation (and their corresponding multiples), meaning much simpler maths and therefore reduced jitter and digital artifacts at higher resolutions. While minimal improvements to latency were achieved the x8 still delivers a miniscule 1.1 milliseconds analogue round-trip latency at the 96kHz sample rate. Even more impressively, latency is still 1.1ms when four plug-ins are instanced on a Console input channel. The UAD plugs generate zero additional latency on the input even under heavy DSP processing loads. SURROUND BOUND

AV post-production is an area where the Apollos haven’t really had a big impact but that may be about to change. The Apollo x6 offers 5.1 monitoring and all the other units offer up to 7.1 surround. The surround software is still in the final stages of development and is due to be released in the next few months. As I write this details are still a little thin on the ground but the gist is you will be able to setup a 5.1 or 7.1 monitoring matrix via the software in the UA Console application and assign it to the physical outputs. There won’t be surround versions of the UAD plug-ins per se (which is a bit AT 43


CENTURY TUBE CHANNEL STRIP Universal Audio’s own plug-in Century Tube Channel Strip diverges from its typical path of meticulously digital modelling classic hardware. While utilising a tube topology in the preamp stage and an opto-style limiter for dynamic control, it has no historical progenitor in the analogue hardware world. The Century is optimised to work hand in hand with UA’s Unison mic preamp modelling system, and as such this plug-in is very much aimed at the tracking stage of the recording process and offers a streamlined set of controls for quick and easy tonal and dynamic massaging on the way into your DAW. The Tube Preamp section sports the usual controls with the high pass filter being noteworthy for its smoothness and usability. The EQ has fixed high and low bands at 10kHz and 110Hz respectively, while the midrange offers a large sweepable range between 300Hz and 7.2kHz making it a very versatile tool. The Opto Leveler is a simple one-knob affair and does a nice job of smoothing out transients and adding beef to sources. Finally there’s a simple output control and these last two stages have informative VU-style metering. Overall, I found this plug-in to be a very handy tool indeed and quickly grew to like the no-frills grey presentation and the simple array of controls on offer. I

of a shame) but, as is already the case, they can be used in linked multi-mono mode within DAWs that support this approach. This looks like a bit of a toe-in-the-water move by Universal Audio and they are on record as saying future developments in this area will be driven by customer needs so watch this space. X MARKS THE SPOT

After checking out the specs and playing with the talkback mic and a few software parameters, my first proper experience with the Apollo x8 was a simple acoustic guitar and vocal recording. I hooked the unit up via a Thunderbolt cable (not supplied) to a Mac Pro laptop running Pro Tools, downloaded the Apollo software and was quickly in business. I plugged a Neumann U87 into the Apollo’s built-in mic preamp and placed it about a foot and a half away from the sound-hole of my Martin steel string acoustic. The Martin sounded sweet and balanced with the Apollo capturing every nuance of string and wood beautifully. Having been an Apollo user for many years the sound of the new unit was very familiar, with perhaps a hint of extra definition and dimension to the sound. I added some layered vocals and a few extra guitar parts to hear how the tones stacked up and found the character of the new converters very pleasing indeed. The next session was with Melbourne synth-pop act the Sugar Glass Project. We tracked electric bass, drums, acoustic and electric guitars plus a rough guide vocal through the Apollo x8 via a variety of mics and preamps. The results were great and I enjoyed using the built-in talkback mic (having assigned it to the multi-purpose Function button on the front panel). It was a vibey session and the band went home happy. With one-off sessions like this it is hard to rate the new converters against the old as there AT 44

tracked a full acoustic-based song using just the Century Channel Strip with Neumann U87 and SM57 microphones and was really happy with the results. Acoustic guitars gained body and shine with some EQ and subtle compression, vocals sounded sweet and present and drums also worked nicely with the Opto Leveler giving me some nice grit and density on overheads and bass in particular. The preamp sound is subtly flattering without adding too much hype or mojo when you don’t want it. Signals can get pretty furry when driven hard through this ‘circuit’ and its not necessarily a pretty sound so you need to watch your input levels. The EQ is basic but very useable while the leveller keeps things nicely in check. The name of the game here is gentle signal optimisation while tracking, so kissing your sources lightly with EQ and compression is the way to go rather than sticking the tongue in (so to speak). The leveller can get a bit grabby above -3dB of compression and too much boosting of those set EQ frequencies across a bunch of tracks will create problems down the line, but if you keep things light it’s a great way to sweeten and groom tracks for an easier ride come mix time. Full marks to UA for coming up with a tasty tube-based design that really takes full advantage of the Unison software/hardware interface.

are always so many variables on any given day of tracking, but I was really happy with the sound of the new Apollo. To better assess the character of its converters I did some A/B tests that gave me the chance to sit back and listen more objectively. TESTING TIMES

First up I looped a longish passage with electric guitar and keys via my loop pedal into a guitar amp and set up an SM57 right on the cab and a Neumann U87 as a middle distance room mic. The mics were recorded into Pro Tools via channels 1 and 2 of the Apollo x8 as well as the same channels in an original silver face Apollo with identical gain settings. I then overdubbed a simple drum part over the top with the same two mics on overhead and room snare duties. A second test involved one of the most important conversion tasks in my studio — transferring multi-track tape recordings into my DAW. For this test I used an excerpt of a simple vocal, guitar and drum machine recording by South Australian singer-songwriter Louise Adams. Again both Apollo machines were used with identical gain settings on the line inputs and all recordings were done at 24-bit/48kHz resolution. Finally, I recorded a mastered track by Sydney artist Emma Davis through the line inputs to hear how the converters ‘heard’ a more polished final product. Repeated listening through various sets of speakers and headphones revealed some interesting differences. Most noticeable to me was a subtle tightening of lower-mid and bass frequencies in the x8 recordings. In general the bottom half of the spectrum sounded a little more defined and well articulated. The new Apollo also exhibited a slightly different character in the top end with a tickle more sheen above 10kHz and improved management of things like sibilance around the critical

4-8kHz zone. These differences were far from pronounced. They were subtle but noticeable with careful listening and more or less in line with my expectations of the x8. At this level, enhancements are incremental and unlikely to reveal night and day differences. The Apollo x8 sound shares many qualities with its older sibling but to my ears the fidelity and realism of the sounds has gone up a couple of notches. X FACTOR

Universal Audio has once again positioned itself well in a crowded marketplace by updating the Apollo converters and adding value to the new X range. Whether the incremental improvements in sonics, processing power and features are enough to see a mass migration from the older units to the new ones remains to be seen but initial preorders are strong in Australia and the units have been very well received overseas. Expect further software releases from UA in the near future that will unleash the full power of the X range with surround monitoring being the next cab off the rank. Who knows, perhaps we’ll see UAD-3 plugins before too long as well. Those who do pull the trigger on an Apollo X model will certainly not be disappointed by the sound or versatility of the units but, as always, buyers should be sure to keep a few extra dollars stashed away in the cookie tin for all those tasty UAD plug-ins.

Hear the Difference To check out Greg Walker’s comparison between his original Silver Face converters and those on the x8, head to soundcloud.com/audiotechnology


AT 45


REVIEW

ALLEN & HEATH SQ SERIES Digital Console

The SQ is less like a Qu upgrade, and more like getting a chunk of d-Live for a steal. Review: Mark Davie

NEED TO KNOW

When Allen & Heath entered the Battle of the 32s with its Qu-32, it did so in a real Allen & Heath way. Behringer had made its play with the X32 — stuffing every ounce of processing into a package fronted by a crowded screen. Then Behringer’s now-sister company, Midas retooled much of that tech with a ‘Bentley-designed’ chassis that didn’t make huge strides in operability, nor did it turn it into a ‘real’ Midas. All of the 32s were ‘bridging’ consoles. Fully digital desks, with a full complement of onboard analogue I/O. That way, anyone with existing analogue infrastructure didn’t have to rewire a single cable, drop box, or core if they didn’t want to. They could simply replace their analogue console with a digital one, and lose a rack of outboard gear in the process. Knowing this, Allen & Heath straddled this analogue/digital divide on the Qu-32 surface side, too. You still had to get out the white ‘lecky’ and chinagraph to label up your channels, it

PRICE Consoles 16+1 fader SQ-5: $5499 24+1 fader SQ-6: $6499 32+1 fader SQ-7: $7999 DX168 16-in/8-out Stage Box: $2999 AT 46

CONTACT TAG: (02) 9519 0900 or info@tag.com.au

had big buttons on its touchscreen, loads of colour, and there weren’t too many layers to get your head around. If you came from an analogue console background, this was about as ‘at home’ as you could feel on digital. NEXT IN QU-EUE

The Qu series was a raging success; not only was it easy to use, but it had plenty of clarity, too. In the meantime, Allen & Heath overhauled the top end of its digital range with the D-Live series; introducing 96k sampling rates, almost doubling the channel count of iLive, and building more professional interfaces and touring packages. It also split the range into two surface ranges; the touring-spec S-Class, and the install/smaller rental house-spec C-Class, which comes with a few less knobs and buttons, but still operates the same engine and racks. There was, of course, a huge gap in the middle.

PROS Flagship DSP processing onboard Interfaces with dLive & Qu gear Sounds punchy & clear Completely flexible routing & layout

CONS MADI not available yet

A console that could still drop-in as a replacement for an analogue console, but make use of some of that d-Live tech. The SQ series is it. Like the Qu series: each console has a healthy collection of I/O on the back; the processing engine is onboard (not housed in the I/O rack); and it has a similarly user-friendly interface. On the flipside, like the d-Live: SQ runs natively at 96k; the channel and bus-processing count doesn’t change with the size of your surface (you can operate the 16+1 fader SQ-5 surface and still have the same 48-channel/36-bus architecture as the 32+1 fader SQ-7); it’s got multi-colour LCD scribble strips; fully-customisable fader banks; an LED light bar; as well as dLive multi-colour knobs, and customisable soft rotaries.

SUMMARY There’s not much the SQ hasn’t taken from the flagship dLive system; same 96k processing and latency, same engine core, can use the same preamps. The only thing it’s lower on is screen real estate and size… one of those is a bonus.


UPGRADE PATH

Besides the different core architecture, Allen & Heath has upgraded every key area of the SQ, when compared with the Qu. There are now eight onboard FX engines, instead of four. However, only the first four have dedicated FX sends as well as returns. The last four have dedicated returns, but you have to give up one of the 12 auxiliaries to access the other four, or use them as inserts. The touchscreen size remains the same seven inches, regardless of the console size. The onboard USB recording and playback has gone up from 18 channels to 32. The mute and DCA groups have been doubled. The 32-fader surface has 16 soft keys instead of 10, as well as eight soft encoders. The fader layers have been upped from three to six, and they’re all now fully customisable thanks to those LCD scribble strips. One change some may not prefer is the fewer number of encoders dedicated to the parametric EQ section. Rather than discrete Gain, Frequency and Width knobs for each of the four bands, there’s only one set, with buttons to switch bands. Also, while Virtual Soundcheck can be implemented, it’s limited to the 32 x 32 SQ-Drive recording and playback; you don’t have access to the full 48 channels. On the connectivity side, Dante is a great addition and the flexibility of S-Link is excellent if you’re mixing and matching Allen & Heath consoles. However, the lack of any MADI integration or an optional AES expansion card might be limiting for some scenarios. Given all those protocols are available as option cards on the d-Live, we might see them filter down to SQ, as well. FIRING IT UP

Having gotten along well with the Qu, I was eager to boot up the latest from Allen & Heath. While the SQ series can interface with the 48k AR series of stage box extenders, TAG supplied me with a 96k 16-in/8-out DX stage box. At our local church, I ripped out our trusty Digico SD9 and D-Rack, and hooked up the 32+1 fader SQ7 and DX rack. Being used to the soft power delivery of the SD9, I’ve become a little lazy with my boot up order. Obviously, amps should be turned down when you power up your desk, or powered up last. They weren’t in this instance and I got a bit of a fright when the SQ7 fired up with a bang. The next time, I was more prepared. The first thing that became apparent was the size. The SQ7 was almost half the depth of the SD9 (which has dLive-sized touchscreens), making it look like a child swimming in the oversized pants of our AV desk. You could save a lot of room with this console; it’s smartly packaged. Operating on it isn’t at all claustrophobic. Losing the full set of EQ knobs frees up a lot of room around the screen, and the symmetrical knob layout feels spacious. As far as I’m concerned, the more colours the better. I never feel like I suffer from over saturation. I’d argue we could use even more differentiation within the major hues. The new d-Live knobs are spectacular in that regard. The central LED illuminates in a colour specific to the function,

THE RACK At present, there are two 96k stage boxes available for the SQ series — the DX168 and DX32 expander rack with four available I/O slots. I was sent the DX168, a 16-in, 8-out unit. Because it was built as an extender for the dLive system, it has the same preamps as the main rack. There are marginal spec differences between the stage box preamps and those on the SQ console itself, and they also have the same gain staging. In its natural form — like Allen & Heath’s 48k AB128 — it comes with a metal handle at one end, with all corners protected by chunky rubber bumpers. It can also be rack mounted with an optional kit. There are red phantom power LED indicators located next to each input XLR, and green LEDs for power and connection status. There’s no ‘on/off’ switch, and it’s powered via a standard IEC power

cable. There are two DX Link Ethercon ports on the side. For a standard single-box operation, simply plug into DX Link A. You can easily expand your remote stage box I/O count to 32-in/16-out by linking the DX Link B port to an A port on another DX168, and switching it into Cascade mode. The other mode is a Redundant mode, but is only applicable to the d-Live series for now, no word on whether Allen & Heath will add redundant looping when an optional S-Link card is added. If you want to add the full 48 input channels via remotes stage boxes, you’ll have to add the optional DX Hub. It was designed to split the d-Live’s gigaAce protocol into four 32 x 32 DX Link ports, but you can also use it to split the SQ series’ S-Link protocol into four DX Link ports.

MONITORING NEW FEATURES While reviewing the SQ7, it was up to firmware version 1.2. Just recently, Allen & Heath released a 1.3 update. It was too late to really test out the new features, but they’re worth mentioning. Allen & Heath has spruiked the SQ7 as a do-it-all console; equally at home at monitors — with its 12 stereo mixes for in-ears — as it is for FOH. However, there was no dedicated level control for an engineer’s cue wedge. The new firmware will allow monitor engineers to assign the master fader to that role.

which is of great importance when using the eight customisable soft rotaries. These are defined by a three-layer system. You start with the Function, e.g. Send Level Fader, or Compressor; then you specify the Channel, e.g. Kick; then the specific Destination of Parameter, e.g. Aux 1, or Threshold. However, in the above example, the display will show ‘Kick’ for both rotaries. It’s the colour LED on the knob that gives away the Function, e.g. light blue for Sends, and orange for Compression. It could do with a third level of detail in those cases, but it’s better to have them than not. FAMILIAR LOOK

The Allen & Heath digital look is well and truly alive. If you’re familiar with a Qu, you’ll have no trouble jumping onto an SQ. Big buttons in wellspaced layouts, and a plethora of options made simple to navigate. Your local and remote I/O is all available in tabs on the Routing page. Within each tab is a matrix, allowing you to assign, for instance, Input 1 on your DX168 stage box to Input 13 on your console. Your sources are laid out along the top of the matrix, with console input destinations down the side. Outputs are on a separate page, keeping routing nice and tidy. With the matrix, it’s also easy to send the same stream to multiple physical outputs. I sent the Master LR mix to a foyer send with zero effort. Once you’ve routed all your I/O, it’s time to

This lack of monitoring control was also apparent on the FOH side, where the only outputs for the PAFL were the headphone output and the RTA. Occasionally you want to PAFL to your main system, or at least a pair of monitors on your bridge. This has been addressed in the latest revision. There have also been additions to what you can add to your scenes (snapshots), as well as the ability to save your own channel processing libraries. A big one is the addition of more plug-in choices from the d-Live stable, available for purchase.

customise those fader layers. In the setup section, there are two lanes. The top one contains your available inputs and busses, and the bottom lane shows your selected fader layer. It’s as easy as dragging ’n’ dropping any available channel — whether an input, FX, aux, etc — to the fader layer below. If you want to leave a gap or delete a channel from your fader bank, simply flick it away with your finger and it will disappear from the layer. DCA assignment is similarly easy, all done from a single screen. Just select your DCA and hit the Routing button to add and subtract channels. Ganging channels into stereo pairs is done on the Mixer Config page, where you can switch pairs into either mono or stereo. When in stereo, it combines them onto a single fader. There are also loads of options for stereo imaging, whether its just switching the left and right, or phase flipping one side, mono-ing the feed by sending both left and right to each side, or decoding M/S streams to output M+S/M-S. While you’re assigning channels, you can also set up a full 48-channel auto mic mixing setup. Either in two separate groups of 24 channels, or one huge auto mix. It’s not a feature I often use, but the absolute right tool for certain scenarios. It’s a huge addition, especially given it’s part of the basic package. PROCESSING CHAIN LINK

Each channel has a typical processing strip. At the top is the preamp section, with a gain knob, AT 47


meter and the usual suspects like +48V phantom (hold down to turn on) and phase flip. Gain goes from 0-60dB in 1dB steps, with a digital trim of ±24dB for plenty of flexibility when gain sharing for monitors. Allen & Heath is also giving away a tube-style preamp emulation to add a bit of drive to your input stage. Next is a readout for the fixed-slope HPF, which is also displayed in the parametric EQ section. After that is a functional gate section, which was dead easy to operate. It has your typical attack, hold and release parameters for dialling in a naturalsounding response. Threshold and depth (0-60dB) controls complete the gain reduction picture. As well as red meter showing its action, there’s also a 12-second histogram graphing out its signal reduction in realtime. You can also side-chain the gate to another channel, as well as engage a highpass, band-pass or low-pass filter for your key. Next in line is the insert, which can be fed to and from any number of analogue and digital I/O, including one of the internal FX engines. There’s an operating level option, and the ability to turn the insert on or off, but wet/dry mix is left to your effect unit. The four-band parametric EQ is also quite smart. The physical knobs give you broad control over frequency settings, and you can use the Touch ’n’ Turn knob to more finely dial them in. Despite the broader control when using the main knobs, the frequency choices do smartly snap to ratios and octaves of the other bands. Often when cutting out a problematic frequency, there can be a resonant harmonic in the octave above; this feature makes it fast to get at. You can only access the shelf settings in the high and low bands by winding the Q all the way out with the Touch ’n’ Turn knob. This keeps users from accidentally turning a bell into a shelf when trying to widen a band. The last processor before the pan control is the compressor. The only thing it’s lacking is an auto makeup gain control, and it’s also not multi-band. Other than that, it’s richly appointed for an onboard compressor. You can choose between peak and RMS detection modes, hard and soft knee, a side chain, and filter settings including a bandpass for use as a de-esser. While you can drop the filter in and out, it would be handy to have a key listen feature, especially when trying to dial in a de-esser setting. A big feature is the parallel compression path, with independent level setting for the dry and wet signals. While it takes a bit longer to set than a single Wet/Dry control, it does allow you a different perspective when you’re able to leave your dry signal at full level while dialling in the wet, rather than chasing a ratio. It’s much of a muchness, just more control. If you need more processing power, Allen & Heath has also announced an optional Waves Soundgrid card, for 64 x 64-channel processing. GRAPHIC METERING

On the mix bus masters, you lose the gate, but get a 31-band graphic equaliser, as well as the parametric EQ. There’s a flip fader to GEQ button on the console for easy access. There’s no 0dB detent on AT 48

TALKING ALL TALKS Admirably, while Allen & Heath seems to debut a new Ethernet connectivity protocol with each new series of consoles, the SQ consolespecific S-Link protocol speaks all of the other Allen & Heath languages, too. It has no trouble conversing in dSnake with the 48k AR series of stage racks. It can even plumb directly into a Qu console. Likewise, at the other end, it can attach to a dLive’s gigaACE network. There is one S-Link port on the back of each SQ console, but you can add another with the option-

the fader (either for the GEQ or in normal fader mode), but the select LED lights up green when you’re at 0dB, and you can hit the select button to zero out the fader. There’s an onboard 31-band RTA, which follows the PAFL selection. If you plug in a measurement mic, it can help tune a room, with prominent frequencies displayed in red. There’s plenty of onscreen bar graph level metering throughout the console screens, including a dedicated Input Meters screen and a small meter attached to every processing section of your channel. There’s also a multi-segment LED meter that follows your PAFL or defaults to the main LR mix. For physical monitoring at the fader, Allen & Heath has opted for two LEDs. There’s a red Peak LED, and a multi-colour LED for Signal level. You can customise the dB levels at which different colours come on, or stick to the defaults which will light up from green-y blue, to green, yellow, and orange, depending on how hot your signal is. DOWN TO THE MIX

Mixing is a pleasure on the SQ, and yards ahead of the Qu thanks to the customisable fader banks and LCD screens. You can easily access the four dedicated FX sends and each of your 12 mixes (which can be configured as groups or auxes, pre or post) by hitting the buttons spaced down the side. While there’s no LCD screens to name your mixes, there’s enough space to lay down some electrical tape and label them up. The Main LR mix button is indented from the rest of the mix button row, unfortunately the button is the same colour and size as the rest. With so many available mixes, it’s not hard to adjust the wrong mix layer when you’re finding your way around the console. A bigger button, with a different coloured LED would have helped. A really handy feature is the ‘CH To All Mix’ button. With a channel selected, holding this button down attentions all its associated auxes, groups

al S-Link Card. It opens up all manner of setup schemes. You can pull in an extra 128 channels from a dLive gigaACE network, run three DX168 stage boxes off one port and an ME-U personal monitoring distributor off the other. You are still limited to a total of 48 input processing channels, but it does mean you can get them from more places. The best part is any 48k Qu or AR add-ons are all upsampled to 96k; the native sample rate of the SQ.

and effects into one fader layer, so you can adjust the level of each in one place. It’s momentary, not latching, so you have to hold it down with one hand while you adjust the faders with your free hand. Getting around the console is dead easy, and the onboard effects sound stellar. There was the occasional parameter I couldn’t assign to a soft button or rotary — mostly to do with FX — but they were arguably more scene-specific controls anyway. Overall, the console sounds great. The punchier top end was immediately noticeable after switching out the SD9, which runs the D-Rack stage box. The Audix D6 has a reputation for being a quintessential modern scooped kick mic, but I always found myself winding in more top end click on the SD9. On the SQ7, I ran it flat. Same kick drum, same mic, totally different results. Likewise, a typically muddy rhythm guitar sound cut through with ease on the SQ. I was shocked by how different the two systems sounded, and grappled with the differences during my first mix — feeling like I’d lost a little of the relaxed glue I was used to — but settled in from then on. One of the benefits of Digico systems is their ability to musically handle clipping. Turning it into a pleasing saturation rather than a blemish. The SQ7 was forgiving in this regard, too. The SQ series is really well thought out, and while it looks like the next step up from a Qu, it’s really much more like getting a huge chunk of the d-Live system for a steal. The SQ series can be coupled to a d-Live-derived stage box. Sure, it’s not the big DM48 touring rack with all the redundancy features, but it has the same pres. Onboard, the SQ has the same XCVI core with 96-bit accumulator headroom as the d-Live, the same <0.7ms latency, the same 96k internal processing, it also has the same FX, the same DEEP processing preamp capabilities and will have the same plug-ins available. You’re getting flagship sound in a package that starts under $8k (SQ-5 and one DX168).


7

ART 7

STILL THE FIRST CHOICE OF AUDIO PROFESSIONALS THE WORLD OVER, RCF INTRODUCES THE ART 7 MK4 SERIES. THERE HAS NEVER BEEN A BETTER TIME TO HEAR THE DIFFERENCE FOR YOURSELF. PRODUCTION GRADE BUILD QUALITY. SEVERAL SYSTEM CONFIGURATIONS (UP TO 4” VC). PROPRIETARY FIR-PHASE TECHNOLOGY. FIVE YEAR AUSTRALIAN WARRANTY. MADE IN ITALY.

TO FIND YOUR NEAREST RCF PROFESSIONAL DEALER, VISIT: WWW.RCFAUDIO.COM.AU

AT 49


REVIEW

ARTURIA DRUMBRUTE IMPACT Analogue Drum Machine

Arturia’s Drumbrute Impact may not have wooden end cheeks, but its distortion and ‘Color’ function make those look like elbow patches on a professor’s tweed jacket. Review: Brad Watts

NEED TO KNOW

There was a time when Arturia was steadfastly a software manufacturer, producing some of the finest virtual reincarnations of history’s greatest analogue instruments. Arturia’s V-Collection kicked off with classic models like the Yamaha CS80, Roland Jupiter-8, and MiniMoog, growing it into the collection of 21 vintage synths and keyboards it is today. The V-Collection, for me and countless others, was the turning point for selling off various hardware units. Founded in 1999, it took around 10 years before the innovative French outfit moved into the world of hardware

PRICE Expect to pay $399 CONTACT CMI Music & Audio: (03) 9315 2244 or sales@cmi.com.au

AT 50

PROS The kick The FM Drum The distortion The price

synthesis, touting its Origin hardware synthesiser as the mother of all synths — and rightly so, there wasn’t much you couldn’t pull off with the Origin. BRUTAL KICK OFF

Since 2009 the company has released a slew of keyboard and trigger-pad style controllers, all the while expanding its stable of software virtual instruments and effects. Come 2012, Arturia embarked upon the ‘Brute’ lineage of hardware synthesisers. The series has given rise to a number of synths, exclusively analogue, from the MicroBrute through to the MatrixBrute, and all

CONS None at this dollar

adhering to Arturia’s fanaticism with analogue sound creation and tactile, hands-on control. During 2016 Arturia added a drum machine to the Brute lineup. The DrumBrute offered 17 entirely analogue drum sounds consisting of two kicks, snare, clap, open and closed hi-hats, high and low toms and congas, maraca, rimshot, clave, a tambourine, ‘zap’, and cymbal (both forward and reverse). Like all the Brute range there were no menu driven parameters. Everything is accessible on the DrumBrute via dedicated pots, pads, and controllers, and with enough connectivity to keep any serial analogue synthesist gleeful —

SUMMARY Arturia’s DrumBrute was all about classic analogue drum synthesis done well, while the smaller, cheaper, more characterful DrumBrute Impact adds loads of ‘Color’ and distortion to shape itself as a potential classic.


both in terms of individual analogue outputs and multiple sync options. Subsequently the DrumBrute has forged a sizeable following. In keeping with die-hard analogue aesthetics it even has timber end-cheeks! Recently, Arturia announced another drum machine priced well below that of the original DrumBrute. DrumBrute Impact. While the Impact seems a lesser animal than DrumBrute with a mere 10 drum sounds, less individual outputs, and no timber end-cheeks, DrumBrute Impact is a weapon in its own right. DrumBrute users will no doubt wish to add an Impact alongside their DrumBrute, and those not wishing to stump-up for the more expensive version won’t feel the slightest remorse. But before I get to why, let’s look around at what the DrumBrute Impact has on offer. SMALL FOOTPRINT, BIG IMPACT

Firstly the Impact is smaller. Covering about an A4 footprint you’ll find it easier to both lug to gigs and divvy out studio space for. There’s also only four individual outputs as opposed to 12 on the DrumBrute. These take care of the main instruments: kick, snares, hi-hats, and the wildcard FM Drum. More on that one shortly. For multi-tracking these are a musthave. It’s unfortunate, however, that the other instruments can’t be shunted to these outputs via software. MIDI I/O is the same as the DrumBrute, as is the clock I/O. Clock in and out can be set to cover multiple voltage systems, and the unit can sync as master or slave via USB and MIDI. The sequencer is based entirely on the original DrumBrute, and is quite a revered system. Based on the tried-and-true 16-step model with the ability to hold 64 steps, it includes all the performance features found in the DrumBrute such as roll, randomness level, and the ever-joyous swing amount — accessed via a good old dedicated pot, of course. You can also bump the unit into record and input live drum-hits via the eight pads. The sequencing system is for all intents identical to the DrumBrute apart from one additional feature. All the sounds can be augmented with a ‘Color’ layer. This aspect will alter the sound of an instrument and ‘Color’ events can be added into the sequenced pattern — similar to when adding ‘Accent’ to an instrument and event on a TR-909 or 808. Each instrument has its own slight variation of Color — the kick gets a little distortion, the snares change in pitch, and the cymbals alter in pitch and are slightly excited, the hats have their decay altered, and so forth. With a unique alteration available for each instrument, the palette of sounds available increases exponentially. KICK TO THE FM

balance between tone and snap. The second snare is a lighter snare with adjustment of the overall tone and a pot for decay. The high and low toms share a single pad with a button to switch between the two, along with a pot for overall pitch. Instigating ‘Color’ on the toms will lengthen the decay. The hats are very 909ish and are scrumptiously crisp. The closed can be adjusted for tone while the open offers a pot for decay. To the far right of the eight pads is the FM Drum instrument. A brilliant little instrument with four dedicated pots affecting carrier pitch, modulation pitch, the FM amount, and overall decay. Instigating ‘Color’ on this drum aligns the pitch envelope with the decay. FM Drum is wildly versatile. You can get it to make the most ridiculous noises. From high pitched warbles, repetitive pseudo bass-lines, through to Giorgio Moroder toms (think I Feel Love by Donna Summer), all the way down to longwinded Roland TR-808 kick drums and hokey CR-78 bops and boops. The FM Drum is nuts, and stupidly fun to use. Easy to sample from for a quick bass sound if you’re so inclined. There’s also a cowbell sound you can call on if you must. Reminiscent of the TR-808 cowbell, there are literally no adjustments for this instrument. That’s it. Cowbell. Go for it. Now there are two final features I’ll mention. First up is the to-die-for kick drum. Seriously, go and have a listen to this thing as soon as you get the chance. The decay on this runs out to around three seconds, and the pitch goes down to hertz my monitors weren’t capable of serving up. Simply glorious. Pitch and decay are the only parameters other than adding a little distortion when adding ‘Color’. Plenty of thwack on the attack, and a gorgeous decay. By itself you could pull any electro kick you could possibly require. In fact, you may decide the kick drum alone is worth the admission price. Now add the global distortion available on the DrumBrute Impact and you’ll add hundreds more variations to that kick. The combination of these two is simply gorgeous. Yeah – that distortion circuit I’ve not mentioned until just now really sets the Impact ablaze. The original DrumBrute offered a global high-pass filter with resonance, which is all well and good, but this is so much more! NO NEED FOR EMULATION

Considering the street-price on the DrumBrute Impact, I’m certain the machine will be a runaway success, simply because of the kick and the distortion effect. Then the FM Drum adds a world of versatility to the device again. Go have a listen, absorb a slight incursion on the credit card and grab what will outrun the original DrumBrute as a true classic.

Now to get to what’s really exciting and new about the DrumBrute Impact – the sounds themselves. The first snare is your typical 909 style, with a pot for decay and another for the AT 51


REGULARS

LAST WORD with

Kenton Forsythe, EAW Co-founder

I was at UC Berkeley studying regional and city planning with a side in traffic engineering. Like a lot of students, I didn’t have any money. I was really interested in hi-fi consumer audio and helping people with their stereo systems, making some money on the side. I attended a day of the Woodstock festival in 1969, and that got me really interested in live sound. After graduating I got a job. It wasn’t particularly exciting. I met some guys in a local band. Maybe we could build some speakers for their P.A.? That was myself and Ken Berger [EAW and VUE Audiotechnik co-founder]. The first product we built was the SR215 under the Forsythe Audio name. Ken and I had a third partner. That didn’t work out, so we set up Eastern Acoustic Works in 1978. Our first product was a bass box and horn, which we sold to pro sound retailers and to some local sound companies. We moved into three-way systems. Then we got a call from Carlo Sound in Nashville, guys by the name of John Logan and Rich Carpenter. They toured with The Oak Ridge Boys, doing 300 dates a year. They wanted an easier way to set up their rig. So we built them an all-in-one box — a double-15, a horn-loaded 12 and an HF unit. We prided ourselves in building good cabinets. If you rap on the cabinet, it wouldn’t echo. Sound was pushed out of the front, not out the back. All the horn designs were simple. I think part of the success was the integration of the systems — using good components to make a good loudspeaker system. We were still at our original manufacturing location when we started building the KF850. We had a local guy do the woodworking in an adjoining wood shop, and he’d push the cabinets next door for us to do the assembly. Once the KF850 started to hit its stride we moved to our current location in Whitinsville, Massachusetts. It’s a big site. The landlord said to just grow into it and he’d increase rent every six months. With the KF850 we were looking for compactness. Which led us to stick the horn into the front of the woofer. Which wasn’t necessarily the perfect solution — there’s wraparound energy that affects the high frequency — so we had to deal with that. The result was a cab you could move around easily and was easy to use. You could say there was too much overlap in the coverage when splayed, but people loved it. It got picked up by some significant tours and became the most AT 52

popular PA cabinet going around into the ’90s. EAW expanded really quickly… and it wasn’t all KF850s. We were really strong in the installation market. At one point we had half of all the major league baseball stadiums in The States as well as lots of NFL, hockey and basketball arenas. We still remain strong in the sector. Business was pretty frenetic. At one point we sent somebody over to sit on the production line at [transducer manufacturer] B&C to make sure all the three-inch coil drivers came to us. ‘Don’t send them anywhere else! We need them as fast you build them!’ EAW was known as a company that would listen and solve your problems. L-Acoustics started to make a real impact with V-DOSC in the late ’90s. We were late to the game because we were so dominant, busy keeping up with demand. The KF760 and 730 were our initial responses to line array and they both had birthing pains. So we lost our preeminence. We tried using technologies that differentiated ourselves when we probably should have just built something everyone could understand. In some ways the same thing is happening with Anya and Anna. People think our Adaptive technology is ‘magic’ and they’re intimidated. In the early 2000s we were bought by Mackie. The fundamental mistake we made back then was to move all our manufacturing to China. All anyone could see was the cost savings. They lost sight of something more important: being a reliable supplier. Saying that, after our original contract manufacturer collapsed we partnered with a Singapore company with factories in China and that company has been amazingly reliable. On September 4 this year, EAW was bought by Arturo Vicari, owner of RCF. It’s the best thing to happen to EAW since its founding. Twenty years ago RCF, like EAW, was acquired by Mackie Designs. Then Mackie took them to bankruptcy in 2004. It was the best thing that could have happened. Right now RCF is a much bigger company than Mackie Designs or Loud ever was, with the same people running it. Arturo is an entrepreneurial guy who wants to support the business. He’s got a very good feel for the market. He wants to see EAW expand into the high end of the market. All products coming from EAW will be of high value and high performance.

Kenton co-founded EAW in the 1970s, presided over an era of spectacular growth in the ’80s and ’90s and he remains on the EAW payroll as a consultant and a popular face of the company he started. EAW was recently bought by RCF boss, Arturo Vicari. Pictured is Kenton in the late ’80s with his MS63 and 103 studio systems.

EAW will be developing our adaptive technology further, and announce new products at ISE in February 2019. There will also be some more conventional PA products; that will match the performance of competitors’ boxes twice their price. I’m only supposed to be at EAW two days a week as a consultant. I’m not the decision maker, but I’m there to make suggestions. I like to attend tradeshows and talk about EAW, and people seem to like to talk to me. EAW is now a great place to work. The president T.J. Smith is a great operator. Interestingly, he was a real EAW fan as a kid — he had EAW stickers on his pencil box at school. EAW needs to grow 100 percent to be back to where it was in 2000 when it was sold. That’s a very strong goal. What I learnt from the Loud era, was never let someone with an MBA run a small company. That was our experience. MBAs nearly killed EAW. Arturo brings an entrepreneurial edge. He doesn’t have an MBA. If he needed an MBA, he’d hire one in. Instead he has a lot of experience in the business and has been very successful in his approach. It’s surprising how many people love the idea of EAW making a strong come back.


AT 53


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AudioTechnology App Issue 55  

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