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Editor Mark Davie mark@audiotechnology.com.au Publisher Philip Spencer philip@alchemedia.com.au Editorial Director Christopher Holder chris@audiotechnology.com.au Assistant Editor Preshan John preshan@alchemedia.com.au
Regular Contributors Martin Walker Paul Tingen Guy Harrison Greg Walker Greg Simmons Blair Joscelyne Mark Woods Chris Braun Robert Clark Andrew Bencina Ewan McDonald
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 Miriam Mulcahy mim@alchemedia.com.au
AudioTechnology magazine (ISSN 1440-2432) is published by Alchemedia Publishing Pty Ltd (ABN 34 074 431 628). Contact T: +61 3 5331 4949 E: info@alchemedia.com.au W: www.audiotechnology.com.au (Advertising, Subscriptions) PO Box 6216, Frenchs Forest NSW 2086, Australia. (Editorial)
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All material in this magazine is copyright Š 2017 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. 12/12/2017.
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COVER STORY
Parallel Universe Andrew Scheps Opens Up
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ISSUE 45 CONTENTS
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Reverb on Reverb: Sigur Rรณs Live
How Preamps Quietly Make Big Gains
26
QSC K10.2 Powered Speaker
Studio Focus: Sing Sing South AT 6
Blackmagic Makes Fairlight Free
40
34
Aston Starlight Small Diaphragm Condenser
Chameleon Labs 7603 Channel Strip Review
24
36
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HUNTER CAMPUS TAFE NSW INSTALLATION BY MUSOS CORNER
Musos Corner We recently completed the design and installation of the new post-production audio facility at TAFE NSW Hunter Campus - the first Avid S6 Dolby Atmos training facility in Australia. Featuring Pro Tools HD (with Dolby Atmos), Avid S6 and MTRX, Focusrite Rednet, Focusrite ISA, Neve, SSL, AML, Yamaha - all with full Dante implementation - Argosy, Sterling Modular, and custom built furniture.
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Showroom Open 7 Days. 1 National Park St Newcastle West NSW 2302. PH: 02 4929 2829 www.musoscorner.com.au Flexible payment options available. Ask in-store for detail. AT 7
GENERAL NEWS
MOOG IN THE INNOVATIVE HOUSE Innovative Music welcomes legendary musical instrument company Moog Music to its distribution catalogue. Innovative is no stranger to synths, with brands like Novation, Access Virus, and Elektron in its stable of brands. The arrival of Moog kicks things up a notch. Dr. Bob Moog’s legend grew as he redefined the role of electronic instruments in contemporary music. Moog revolutionised the modern analogue synth with the Minimoog Model D, which is now once again available. Every Moog instrument is crafted in Moog’s Asheville North
Carolina factory. The company’s offerings range from compact powerhouses like the Mother32 all the way to the Voyager XL and reissue legacy Model D and modular systems. Moog products are now in stock at Innovative Music. Get in touch for pricing and availability of the synth you’ve got your eye on. Innovative Music: (03) 9540 0658 or info@innovativemusic.com.au
UNFAIRCHILD 670M II Manufacturer of custom recording consoles and high-end outboard gear, Undertone Audio has launched its re-designed UnFairchild 670M II which was first unveiled at AES LA in October 2016. The Fairchild is a compressor that needs no introduction and it’s coveted by many for its musicality. The compressor’s circuit is beautifully simple — only two transformers and four paralleled vari-mu tubes in the audio path. The Undertone Audio UnFairchild is a painstaking recreation of the classic sound from this circuitry, capturing the best of the past, and now also offering more modern functionality.
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Overcoming the original’s limitations, UnFairchild’s redesigned circuitry improves on the lifespan and reliability of the JJ 6386 tubes, a massive cost-savings for studios and engineers. And the UnFairchild’s focus on flexibility and functionality enables it to handle every compression task on any style of music, from pop to rock to jazz to classical. Eric Valentine’s demo: vimeo.com/243459307
MORE NEWS AT www.audiotechnology.com.au
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| Email: boschcomms@au.bosch.com | www.boschcommunications.com.au • AT 9
LIVE NEWS
CLAIR 1.5AM+ STAGE MONITOR Clair Bros has added a self-powered version on the 1.5AM stage monitor to its One Series line-up. The 1.5AM+ has a 15-inch LF driver paired with a four-inch HF driver, both of which are run by a Powersoft amp module exclusively designed for Clair. The integrated amp delivers two-channel power of 800W and 130W continuous to the LF and HF drivers, respectively. Internal DSP loudspeaker processing gives you Xover, EQ, limiter and four preset options including two 12AM Emulation Mode presets for
a similar frequency response voicing and low-frequency phase response to the 12AM monitor model. The other two presets are designed for low-frequency optimisation on stage for when the 1.5AM+ is coupled with other two-way monitors or stage subs. The high frequency horn in the 1.5AM+ is rotatable (40°H x 60°V standard dispersion) and has flush-mounted side handles for portability.
JBL WIDENS VTX A SERIES Earlier in the year, JBL rejigged its touring system lineup with the introduction of the VTX A series. The A12 was the first cab off the rank and it delivered JBL’s best power-to-weight ratio thus far in an effort to build a box that could handle medium to large touring as well as installs. Now, you can’t always do everything with one box, so JBL has released a wider version of the A12, the A12W, that takes the horizontal dispersion from 90 degrees to 120 degrees. It’s perfect for lower elements in a hang, or when an install needs super wide coverage. Importantly, the AT 10
performance remains largely the same as the original A12. It still holds its coverage pattern down to 250Hz, has a frequency response going from 46Hz to 19kHz, has the same quick rigging, and its dual 12-inch fourth-gen differential driver design provides plenty of wallop. CMI Music & Audio: (03) 9315 2244 or info@cmi.com.au
MORE NEWS AT www.audiotechnology.com.au
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SOFTWARE NEWS
V COLLECTION 6 GETS A NEW 4 When it comes to quality analogue-modelling software synthesizers, Arturia’s V Collection has never disappointed. Besides the impeccable recreations of classic synths and keyboards, we’ve always loved Arturia’s idea of consolidating all those instruments into a central performance-optimised location by creating Analog Lab. Version 6 of V Collection has just been announced, and to everyone’s delight it comes with four brand new recreations — the Buchla Easel, Fairlight CMI, Clavinet, and Yamaha DX7 (still notoriously hard to program). The newbies sound super impressive and the GUIs look so authentic you
kinda wish each one came with its own desktop background image. That takes the total number of instruments to 21 and of course every V Collection instrument can run as a standalone application or DAW plug-in (AU, VST and AAX formats). V Collection 6 also brings updates to two existing instruments; Piano V 2 has three extra piano models, and Analog Lab 3 has a redesigned GUI, new sounds, models, and controller integration. CMI Music & Audio: (03) 9315 2244 or www.cmi.com.au
EASTWEST HOLLYWOOD CHOIRS EastWest has a new 59GB sample library that’ll take your choral arrangements to the next level. Made up of male and female sections, you can pull vocal tones ranging anywhere from angelic to dramatic to majestic to thundering. But Hollywood Choirs has a major trick up its sleeve. Instead of the usual deal where choirs sing borderline-intelligible phrases in Latin, the EastWest instrument has an improved WordBuilder plug-in which lets you type in a phrase of your choice and the software intelligently strings together the correct samples to form the
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sentence. Pair that with a MIDI performance and you have a very convincing choir singing the melody and words of your choosing. Nice. Hollywood Choirs Diamond Edition includes a 13 microphone-user-controllable setup (with a Neumann Dummy Head) designed for surround sound and VR. Hollywood Choirs is available for individual purchase (US$599 for Diamond version, US$499 for Gold version) and as a free addition for ComposerCloud subscribers.
MORE NEWS AT www.audiotechnology.com.au
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FEATURE
Multi-Grammy winning engineer Scheps just moved from LA to the UK, but not much has changed for the ‘in the box’ mixer. It’s still all about the song. Interview: Allon Silove
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Whether its mixing Adele, Green Day and about sitting in front of my Neve. If something Black Sabbath, engineering for The Red Hot wasn’t working in the mix, you could pull all the Chili Peppers or working with producers like Rick faders down but leave all the EQ and the inserts, Rubin, Andrew Scheps’ work has always stood out and then start the balance over again. To do that for being powerful and bold. Scheps isn’t afraid of now, I would need a 100-fader controller, which making big moves. Over the last few years he’s would mean a lot of banking around. That would migrated from being an ‘analogue man’ to mixing just irritate me. I was over that when it came to solely in-the-box. Recently, he moved to the UK mixing. I can do that more easily now in a Pro after 25 years in Los Angeles. AT Tools session with VCAs and I caught up with him to find out almost never pull everything down. more about his current approach to You just mute and then add stuff RDING mixing. back in, one at a time. FAVOURITE RECO “It’s really interesting how little AT: Why do you think your move ENGINEER ER. THOSE has actually changed,” said Scheps to ‘in the box’ caused such a stir in RUDY VAN GELD UE NOTE about the move, noting that it was the industry? ’50S AND ’60S BL RY VE E all business by email and Skype in AS: It was really weird, but RECORDS AR P FROM LA, it’s just a different time zone. seemed to be related to two things. DIFFICULT TO TO GINEER A RECORDING EN “People ask if now I only work with The first was that I had been such an . IVE CT PE PERS UK acts instead of US acts, but no, advocate of mixing analogue, and I it’s exactly the same.” think along the way I said something AudioTechnology: Did you about analogue feeling ‘better’. move all your gear over with you? I meant the hands-on visceral Andrew Scheps: I had been looking for a studio experience of grabbing faders and having knobs to partner with who could take all my gear that I that never changed. had accumulated over the years. It was impossible The second was these strange conspiracy theories to find one in the US. Then Monnow Valley in that I was lying about being completely ‘in-the-box’ Wales came up. I had worked there on and off over in order to sell plug-ins! the years — it was one of the first studios I mixed I think the reality is a lot of people who are and produced at in the UK — and always felt really aspiring to mix identify with people who have comfortable and productive there. They needed been successful and how they work. Then they get to replace their SSL (E series with a G computer, bummed out when the person changes how they I think) and my Neve 8068 Mk II (64 input, 16 do things because it’s easier to believe it’s the gear bus, flying faders — technically, its two of them making it sound good, not the person: ‘If I had a tied together) went in the control room as if the Neve and a wall of outboard gear then I’d also be as control room was built for it. All my outboard good as that person!’ It takes away all the excuses gear, instruments and microphones are there too. I and I guess people don’t like that. I remember always work there if I’m tracking — which doesn’t feeling that way too, which is probably why I ended come up nearly as much lately! The people that run up with so much gear in the first place. the studio are awesome. I would love it if they were AT: Your Pro Tools template is pretty complex, always so busy that I could never work there when I how did you arrive at that layout? need a tracking room. That’s the goal. Even though AS: It started with me trying to be comfortable I mixed through all that equipment for years, it’s and familiar with the tools. As soon as you’re really the perfect desk for recording. Great pres, unfamiliar with your environment, you’re no amazing EQs, it’s flexible, and never breaks. longer mixing; you’re hunting around for stuff! AT: What is your current mixing setup? Therefore, my template basically needed to AS: I’m currently using a MacBook Pro. For be my patch bay. When I was on the console, monitoring I use the HD native card with an there was a ton of gear always hooked up for all HD IO (because the sound of those D to As is the parallel techniques. It would change mix to something I know) through a pair of beat-up mix — I’d either use it or not, turn it down, or Tannoy SRM10s powered by an old Crown DC300. EQ it — but the basic tools were there. It would If I’m working on headphones, its my Sony 7506s only be the individual source tracks that would and a UAD Apollo Twin, because it has a nice big change significantly from one mix to another. For knob and has a decent headphone amp and I’ve just example, when I was first building my template I got an AudioFuse from Arturia that sounds great needed a shared kick/snare compressor to replace for headphones too. Then it’s Pro Tools with an the hardware that I’d used on every mix since Alpha Track controller or Pro Tools Stadium Arcadium (Red Hot Chilli Dock for when I need to ride a fader Peppers). Then there were things FAVOURITE M — which is almost never! Oh, and a like a bass crush that I always had IXING ENGINEER ton of plug-ins! on the console. I never found one AT: You mix solely ‘in the box’ ‘in the box’ that I actually liked, so TCHAD BLAKE IS A HUGE INSPIRATION now, but do you miss analogue I don’t have one in the template. FOR HOW BOLD HE IS. HE mixing? I’m also constantly trying to find HEARS STUFF IN A W AS: I’ve been mixing in the box a limiter that I don’t hate, for the AY THAT IS UNIQUE AMON for about five years and I don’t miss backend of mixes. It’s evolving all G ANYBODY WHO HAS EVER any part of mixing in analogue. the time. I recently opened up a MIXED. THERE’S NOBO DY WHO The visceral experience of grabbing Pro Tools session that was nearly MIXES LIKE HI M. faders and having knobs that never three years old and it was almost changed was something I loved un-recognisable!
AT: Do you have any must-have ‘desert island’ plug-ins? AS: If I’m hearing something that’s not right, I use the EQ3, the Scheps 73, Lo Fi and SansAmp. Those come up all the time. I use a ton of other stuff in my template, but I don’t think of them in the same way. AT: Parallel compression forms a big part of your signature mixing technique, but what makes your process unique? AS: Parallel compression is not a new concept. It’s something that people have been doing for 50 years or more. However, I think that both the amount I use is maybe a little more than most people, and the way I use it is slightly different. Instead of using one parallel compressor for the kick and one for the snare, I use one for kick and snare. I also use three stereo compressors for drums. There’s also my FAVOURIT Rear Bus compressor E PRODUC ER which is usually getting ENO AND LANOIS. TH AT everything in the mix GROUP OF RECORDIN GS except the drums. UP UNTIL MID ’90S W ERE JUST ‘GOD For me, the amount ’ TO ME! TH EY CO U LD the compressors share DO NO WRO NG. is really important. I’ve always tried to understand what’s happening in my mixes and what each element in the signal chain is doing. It might be because I didn’t come up through the traditional studio assistant route. I had no idea how other people mixed. I was a programmer and a Pro Tools guy, I had to figure things out on my own by reading and talking to people. Discovering things for yourself takes much longer, but it meant I really understood the fundamentals. Recently, I was reading about how the famous physicist Richard Feynman learned. He had all these text books that showed established experiments regarding how the world works, but rather than simply accepting them, he re-created them. He wanted to understand why they worked. I did kind of the same thing with my mixing and that’s how I feel about my parallel/ shared compression technique. I did it from scratch and it really made me understand what the mix is doing. It also means I can reach ‘inside’ my mix really quickly when it’s not doing what I want and fix it. AT: You have a reputation for creating loud mixes. Has it always been your way? AS: Yes. When I first started mixing I ended up on a lot of SSLs — which sound good when you push them — or on the Neve 8068 that Rick [Rubin] had in his Hollywood studio, which I mixed 99 Problems on. That board only sounded good with a certain amount of level going through it too. When ISH YOU RECORD YOU W it was quiet it sounded ON D KE OR W D HA limp; when it was too FEARLESS loud, it was distorted. — PINK FLOYD There was this sweet THE DREAMING spot where you were — KATE BUSH really pushing the mix bus, with your master SALISBURY HILL IEL fader at -10dB. — PETER GABR With analogue AT 15
STILL WORK WITH E gear, there’s a sweet spot because lead vocal was always all happen without compromising WHAT WAS IT LIK INDIE ACTS? H… IT you’re dealing with power supplies on fader 24, with W either party. The process is very WORKING N? TH that will start to react differently. the rest of the kit collaborative and by the time SO E MA CK JO R LABEL WORK …MICHAEL JA COVERS ME TO MI Because modelled plug-ins are really and the instruments you are done working with them, . CE X NI INDIE HE WAS REALLY ACTS. I JUST FINISH good now and capture a lot of the in between them. there is nothing about the plug-in ST BE E ED TH LY ABSOLUTE MIXING A RECORD non-linearity, they do kind of the Then there would you feel you compromised on. FOR HAVE OR ER EV I R GE SIN FARAO AND I’LL BE . ITH W same thing. However, with floating be keyboards or With Scheps 73, the concept was K OR W WILL EVER WORKING WITH MY CREATIVE. point architecture in software, gain percussion off to simple; it’s a 1073. However, the LY US LO ICU RID BRIGHTEST DIAMO I ND structure at any particular point the right, if I needed process was surprising. I had a Neve FOR EVERYTHING AGAIN — SHARA NOVA — HIM ON, HE in the signal chain almost doesn’t more than 24 faders. BCM10 with 10 1073s but I didn’t WORKED WITH I MI XE D HER FIRST RECORD ED SONG matter any more. It’s not like the Which is exactly what want to just choose my favourite HAD THE FINISH AT THE SAME TIM THE E AS I OM FR AD HE S IN HI Pro Tools mixer has a sweet spot; it’s I do now, only it’s topone and model it. I wanted it to WAS MIXING STA DIUM G. VERY BEGINNIN just math! What matters is the level to-bottom not left-tobe a 1073, not my 1073. Choosing ARCADIUM (THAT WAS ON THE CONSOLE WH going into a plug-in, particularly right. I don’t ever use the one that sounds ‘different’ ILE HER ALBUM WAS IN TH with dynamics because there are the mix window in isn’t going to work on everything, E BOX). I LOVE TO WORK thresholds, and signals can clip. I do think a lot of Pro Tools. but if I’m sitting in a studio with ON MUSIC I LOVE, SO WHEN the plug-ins I use feel right with a certain amount AT: How important is it that a 1073s, they should all work. They I GET THE OPPORTUNITY, AN of level going through them. That’s just how I hear producer or engineer understands were totally open to make a generic D I CAN AFFORD THE TIME, I WILL it. I get people telling me, ‘just mix quieter and music theory? working-man’s 1073 as part of the ABSOLUTELY DO IT. use the volume knob.’ While that is true, I’m not AS: I started out playing French Artist series. Parallel Particles was relying on hitting the horn, then trumpet; reading lots a more specific concept. I wanted D-to-A converters at a of different clefs. I don’t know if it to do something with parallel certain level as part of should be requisite for every music producer or processing (it’s what I do!) and we couldn’t really my mix chain. It’s the engineer. There are those who are amazing but work out a good way to do it. I wanted it to be …RED HOT CHILI PEPPERS? way everything reacts aren’t musical or can’t talk about music theory and shared compressors but that’s really difficult. It THEM ITH W G to the levels. My mixes arrangement. Also, there are plenty of musicians in would have meant having two plug-ins that need KIN OR W COME TRUE. are actually quieter bands who have no music vocabulary whatsoever. to talk to each other. To its credit, Waves actually WAS A DREAM ORY TO HAVE A HIST now than they were a It’s very important for what I do that I have musical figured out some pretty cool ways to implement D TO FEEL WITH THEM AN year ago, or the year knowledge; being able to count bars, and figure that process, but if you didn’t put all the plug-ins EIR TH LIKE I’M PART OF before that. I guess my out the tempo and time signature. That helps me in the right place it wouldn’t work, which was G WORLD RECORD-MAKIN taste changes too. when working on a song. It’s also about having an not cool. I started thinking about the chains I use ME E SO IS GREAT. THEY’R AT: Where do you appreciation of music. It helps that I listen to tons of on individual instruments, rather than groups. ST MO , ST OF THE NICE start a mix? classical and jazz, even though I don’t work on those I had it all mapped in my head so we went into KNOW. I YS GU ED NT TALE AS: Drums. I know types of records (I would love to do more of them). Waves’ studio in Tel Aviv and programmed most a lot of people will You get a sense of sound design in of the audio processing for the say the right answer a totally different way if you listen plug-in in a day! I wanted a simple is, ‘You always start with the vocal because it’s the to Ravel than if you listen to Eno. four-knob design for the interface, ADVICE FOR most important thing!’ But for me, the vocal is Anything that expands your palette but there was plenty of backlash S? ASPIRING MIXER last. I listen to everything early on so I know what’s and the universe you live in just gives over the look of it. ‘It looks like a BE THE MIXERS NEED TO there but tend to find the drums usually need to you more to draw on. video game!’ My concept — which N T FA SONG’S BIGGES change the most. It’s a big deal for me; if the drums AT: What was it like working may be a flawed one — was that I G ON IT. YOU WHILE WORKIN are driving the track properly, everything else can with Waves on your two Waves’ didn’t want it to look like an audio ST JU U YO CAN’T EVER FEEL slot in around them. The drums will generally have Artist series plug-ins: Scheps 73 and processor; just four knobs that IT. H UG RO TH T HAVE TO GE the biggest problems too as they Parallel Particles? made you listen to what happens THING ME SO X MI T N’ I CA T GE TO VE usually involve the most number AS: It’s great when you turn them so you would HA I I DON’T LIKE. N OF IO of microphones. For some reason RS working with them. make a decision based on what you VE L EA ID THE WHAT IS TONEQU OF THE AKE? people think recording lots of They are super heard, not on how much the needle THE SONG OUT IT’S A TINY BOUT ON’T LIKE IQUE, microphones is a good substitute for smart people. Mike was deflecting. That took a long SPEAKERS. IF I D INDIE LABEL. IT’S THERE IS JUST having just a few microphones that Fradis, the project time but it’s really cool and I think THE SONG, THEN ME. IT WAS STAR N. THERE’S TED TO NO IDEAL VERSIO sound really good. As if by putting manager and leader it gives you a visual version of what PUT OUT THE FIR AN’T ST LOW NO POINT. YOU C up extra mics they’ll eventually for all the Artist you hear. Waves has this incredible ROAR RECORD AN E KICK D IT’S JUST MAKE TH make something usable from it! series plug-ins, has team with a long history and are a A LABOUR OF LO E TH OR , VE. THE DRUM BIGGER I then lay out the mix exactly an amazing ability group of people who are astounding T N’ IDEA WAS TO PU . THAT IS T OUT VOCAL LOUDER the same way I used to work on the to see what both in their work. BE MUSIC THAT WOU TO VE LDN’T MIXING. YOU HA . console; left to right. Kick drum Waves and the artist OTHERWISE COME NG SO E TH BY OUT. EXCITED was always on fader one and the needs, and make it AT 16
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FEATURE
When Sigur Rós went down to a three-piece they decided to really strip it back — no horns, no saws, no strings, just Jonsi’s bow. Story: Mark Davie
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And then there were three. When Kjartan Sveinsson left Sigur Rós in 2013, it seemed implausible that the band could go on as a three-piece without some reinforcements. Early on in their international touring days, the band brought fellow Icelanders Amiina along as a string section-come-random sound generator. The saw solos were a true highlight. Then they added The Horny Brasstards (Brassgat í bala in Icelandic) for the Takk… tour. Even with all the extra help, the keyboardist always looked like the floater in the band — pumping organs, clocking glocks, playing any guitar parts that don’t require a bow — doing anything and everything to fill the space around the bass, drums, vocals and Jonsi’s bowed guitar. The band since released their seventh album Kveikur, which was noticeably heavier than even their industrial-sounding debut Von and longtime final number, Untitled 8 (Popplagio). Still, it sounded like Sigur Rós, with delicate arrangements laid amongst a bed of noise. Onstage at Margaret Court Arena for the band’s Splendour in the Grass sideshow, there were two setups — bass, guitar, drums across the front, and a three-station keyboard riser at the rear. I immediately figured the band had pared the string and horn sections back to a laptop rig, but FOH engineer Ingvar Jónnson set me straight, “No there’s no extra players. They go up there and play the start of the second act. They also have two downstage keyboard positions, one for the drummer and a synth between the bass and guitar positions. Before this they had about 12 people on stage with them. I think they wanted to go back to their roots and try to do it themselves as a threepiece band with some playback.” It’s enough to do a
highlights package, explained Jónnson. Years ago, watching the band live at Hamer Hall, there were so many open mics you could hear the creak of the stage as Jonsi shuffled in place between songs. At Margaret Court, it was a completely different arrangement, with half the mics. “It’s cleaner,” agreed Jónnson. “Last time, next to the keyboard they had vibes, glockenspiels, harmonium and all kinds of stuff. About 20 mics on low-level instruments. “It wasn’t really a problem though, because they were only all used on a handful of songs. I also thought about it like mic bleed; it’s almost part of the song. It’s unnatural if you start heavily muting stuff and switching between an open and closed state. If you use the leak and just make it part of the music, it’s not too much of a problem. It’s like when you do pop with a symphony orchestra. You have the pop loud and then 80 mics on the stage. You somehow have to make it work because it would be ridiculous if you start opening and closing them.” It also helps that Jónsi has a good set of pipes on him, despite mostly delivering a sweet falsetto. “The music is really dynamic and he flows with it,” explained Jónsson. “On the loud parts, he is really loud. Some parts he sings really low, but there’s usually not much going on at the same time, so I don’t have to really raise him.” The current show has about 45 inputs coming off stage, with a few extra channels for playback click tracks at the monitor position. The console package was all Midas, and the crew use two 24-channel Midas DL431 stage box splitters to have separate gain stages for FOH and monitors. “I’m a Midas guy,” said Jónsson. “I like the sound of Midas. How
they make the preamps is better than other brands. “The show is really dynamic. I would say the softest part would probably be about 85dB, and the loudest part could be peaking around 103-105dB. Some songs might have a really soft piano and it’s not tasteful to try and make that loud! It would sound silly. “I’m not pushing the low parts up, I just let them be low and then when they play harder you really get the impact of more level and more expression. In general it’s not really a loud show, but in parts it’s rock ’n’ roll level loud.” Jónsson used to run a Waves package, but abandoned it for the internal Midas processing and a couple of TC Electronics M3000 outboard reverbs; one for vocals and the other for drums. Likewise, his scene recalls are mostly there to get him in the ballpark and change reverb presets via MIDI. “It’s programmed for each song, but it’s mostly reverb changes, delay times, and fader positions for the beginnings to give me a fair start,” he said. “I’m not recalling mid-song and we’re not running time-codes.” While he has compression on every channel, they’re often not very active. He also mostly just high passes and adds little bits of EQ. Likewise, the drum gates are only set to trigger at the level of the stage noise, everything else gets through. “They’re happy working on how they want to sound and I think it’s not my position to change that,” said Jónsson. “I’d rather try to capture the sound and feeling they’re creating and expand it to others without messing with it too much. Not adding my personal taste and flavour into it.” Really, said Jónsson, the Sigur Rós show is all about “reverb, reverb, reverb and then more reverb.”
The Sigur Rós show is all about reverb, reverb, reverb and then more reverb
DOUBLE KICK DRUMS Orr Páll Dyrason’s drum kit has two kick drums. The smaller of the two is his main kick drum and “is more like a rock’n’roll kick,” said Jónsson. “I have a Shure Beta 52 and 91 inside.” He only has a Beta 52 on the big bass drum, because it’s used for an effect bass drum, and he wants to cut some of the boom to be able to get it to sound a little older. From there he has a standard Shure SM57 on snare top and a Beta 98 gooseneck on the bottom. Jónsson likes to angle his top snare mic fairly shallow. “I like it close to the rim to try to get more of the whole snare, including the ring, when he hits it.” He has some Shure SM81s on hi-hat and under the ride, Sennheiser e904 dynamic mics on the toms and a pair of KSM42 large diaphragm condensers for overheads. AT 19
JONSI’S GUITAR Daniel Johnson is Jonsi’s guitar tech and luthier. He mostly works with the Black Keys, being from Akron, Ohio. He’d been teching for Judas Priest, and made them some guitars, when he became friends with Dan Auerbach as the band was coming up. He made Jonsi’s main custom guitar almost 12 years ago, and recently put the finishing touches on a second. It was about time, considering the scars and marks lefts from where he runs the bow over the body, hits the strings with it, and drops the guitar at the end of every set. The guitar was designed with a neck that runs all the way through the middle of the body, purely to withstand that drop. It also has a gap between the bridge and tailpiece so Jonsi can draw the larger squeaks and high harmonics out with his bow. Johnson calls those squeaks ‘magical twinkles’, saying “bowing a guitar is not always perfect, it jumps octaves and an overtone squeaks here and there.” Johnson said the sound behind all of Jonsi’s reverb and distortion is simply a classic rock tone. “The guitar has a real classic Seymour Duncan 59 humbucker,” said Johnson. “It’s not a hot wind, more open with a lower output.” At the other end of the chain the guitar is fed into the clean channel of two Marshall JCM2000 DSL heads wired to a pair of Marshall quad boxes. “If you turn off the reverb with no guitar pedals on, it’s just a very bright, twangy, classic rock sound,” said Johnson. “If you hit it with a pick it will sound kind of AC/DC-ish.” In-between, Jonsi has a Suhr Isoboost full-range booster that runs into a pair of distortions; a Suhr Shiva Drive which has its own mid-range boost, into a TC Electronics Booster+ Line Driver & Distortion pedal. “It has a treble and bass control on it, and the bass is all the way up on it,” said Johnson. “That puts so much low end into the reverb that it starts this tumbling sound.” The signal is then buffered and sent to a stereo pair of TC Electronics M350 reverb units in his rack: “The reverb decay is so wet and long that the fundamental is really buried.” The signal continues on in stereo from there, going through the heads and back out to the cabinets. “His cabinets have different types of speakers in them, old vintage 30s and 75W Celestions, the lower and higher wattage makes them sound a little different,” explained Johnson. They’re also miked by different mics; a Shure KM313 ribbon and Sennheiser e609 dynamic. One on each type of speaker, with the pairing flipped for the other cabinet, so each speaker type is miked by both amps. It’s all about the power of the sound. At first Ingvar put them up to try out different mic/speaker combinations for each song, then he just “ended up using them all,” he said. “It just gets a richer and fuller sound.” The end result is in stereo, but it’s very subtle. “I can hear it because I have one amp in each of my in-ears,” said Johnson. “For me, it’s more to hear if one amp has an issue, whereas Jonsi listens to it that way because it sounds great.”
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ROUND THE WORLD WITH GALILEO Jónsson has been with Sigur Rós on and off since the band released their second album Ágætis Byrjun in 1999, which was also their first major international hit. Besides mixing FOH, Jónsson works as a systems engineer for one of the two main pro audio rental companies in Iceland, Exton. At Exton, they have a ton of Meyer Sound inventory, so Jónsson tours with a Galileo processing unit, which he places over the top of the house processor simply because he’s used to it. He explained the outcome he’s looking for is pretty straightforward. “I’m basically just trying to get the system as flat as I can,” he said. “It usually doesn’t take me very long if the system is properly set up.” He also heavily relies on Smaart. “I can’t do anything without it, it saves me so much time,” said Jónsson. “I’ve been using it since version three. I use it mainly before the show when we’re optimising the system. During the set, I equally look up the frequency response and the impulse response to see any problems. If there’s some resonance in the room or instrument, or some notes are standing out you can see it there.”
d&b ARRAY PROCESSING JPJ Audio system techs, Jesse Mahoney and Christie Daly, have had a busy run of sideshows this Splendour season. They’ve both been jumping between all the major venues, including Festival Hall, where they’ve had an L-Acoustics K1 system to handle bands like Queens of the Stone Age. By the time I get to them, they’ve done about 15 shows straight between them. At Margaret Court Arena, the PA has been a d&b J-Line system, but the rig has been far from set and forget. To make extra room for some last minute ticket sales at the LCD Soundsystem gig the night before, the stage was shrunk by eight feet, which had to go back in for Sigur Rós. “We’ve had to put the PA up and down three times, which requires realigning the system,” said Daly. “There’s been very busy mornings, followed by okay afternoons. Pulling down the PA is about 80% of a load out, we just don’t have to take it out of the room.” The J Line feels as close to perfect for the arena, because there’s a limit to how much weight can be flown from each point, precluding the use of heavier systems like K1. “The J Line is fantastic because it’s a lot of PA in a really light box,” said Mahoney, saying the K2 is also a good fit. “We’ve been told it can handle about a tonne per hang, which is the point loading from the roof. We’ve got 12 d&b J line in each front hang. That, plus the one tonne motors is 900kg.” The PA is also hung a little wider than they would prefer, because they aren’t allowed to hang bridles, “but it’s not enough to cause any problems,” said Mahoney. Overall though, he said Margaret Court Arena is “fantastic, it’s been designed to be very acoustically even. It rings a bit in the low end, but not unpleasantly. I don’t like to add much EQ before handing over a system, they come out of the factory very AT 22
flat, so if you’re hanging them correctly, you shouldn’t have to do much.” JPJ has also moved one step forward in terms of coverage by adopting d&b’s array processing technology. It’s a couple of years old now, but with each box requiring its own cable and amp channel on the new D80 amplifier, it’s been more resource intensive to apply in the field. JPJ had used it for one-off shows like Coldplay, but wanted to use it for the Splendour side show run, “because all the clients coming through are all world-touring bands and as a company, we wanted to give them the best we could possibly provide,” said Mahoney. “You plug the design of the venue and system into the 3D software. The system makes calculations based on your array prediction, and then processes each individual box in the hang to provide very even coverage from the front to the very back. It’s the goal of all sound systems, but the way this technology does it is particularly good. If you walk from the very top seats to the very front, it sounds exactly the same. In a conventionally processed system you’ll always notice a drop off as you walk away from the system. Line arrays have been very good for making up for that, but even so you get a natural drop off of the sound. With array processing, that goes away, and it’s freaky how good it works. Once you start using it, you don’t want to go back. You can punch it in and out for FOH and it’s like night and day. The J Line is a really good system, whatever you do to it, but using array processing makes it go from sounding like it’s coming from speakers side of stage like a normal concert, to sounding like it’s coming from two feet in front of you face. It’s worth it. We’ve got over 100 J Line boxes in touring stock, and we should have enough amps to fully array process every system by the end of the year.”
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BLACKMAGIC BUYS FAIRLIGHT, THEN GIVES IT AWAY Aussie video titan, Blackmagic Design, has made a habit of buying expensive pro software and giving it away. Now you can get Fairlight free of charge. Story: Michael Gissing
Not many active musicians have touched stylus to CRT screen on an original Aussiemade Fairlight CMI. The effect of Fairlight’s ground-breaking sample-based computer music editor is more likely to be felt in modern software workstations — like Ableton Live. Fairlight didn’t become ubiquitous in the music production space, instead the company turned its focus to film and broadcast and made arguably the world’s best DAWs for those industries. In keeping with its history of innovation and survival in changing markets, Fairlight has once again undergone a phoenix-like transformation to be reborn under the watchful eye of another famous Australian name — Blackmagic Design. Blackmagic should be a familiar name to audio folk as the divide between audio and video is dissolving rapidly. It started modestly, making digital video I/O cards for editing systems like Final Cut Pro, and has grown into a juggernaut that dominates NAB — the world’s biggest broadcasting show — each year. As well as developing classy hardware, Blackmagic has been buying up software companies and giving away once exorbitantlypriced film studio software free of charge. The first big purchase was Da Vinci, a company that made colour grading software and hardware. A system that cost six figures was rapidly reengineered into a software-only, cross platform version called Resolve and released free of charge. A US$999 version had some additional features but the free version did everything most users needed. Within a few years the team had advanced Resolve’s editing functionality, added lots of finishing tools like title editors, and embraced audio and video plug-in support. Next to be snapped up was Fusion, AT 24
one of the best VFX compositing tools. Again, within a short time a free and US$999 version hit the stunned market. Naturally, when Fairlight offered its IP to the open market on the 22nd of July, 2016, Blackmagic took a look. A deal was announced on the 9th of September at the start of IBC and speculation ran hot over what Blackmagic would do with the audio company. When you consider how quickly Blackmagic has grown into a complete one-stop shop for video production — including pro software, video conversion and live streaming hardware, even developing its own professional camera range — the pace of development at its competitors appears glacial by comparison. Still, the appearance of Fairlight inside Resolve at NAB 2017 stunned even seasoned post people like me. How did they manage to integrate a professional audio workstation into Resolve in six months? To go behind the story of how two Aussie companies produced a mini miracle and to find out what it all means for existing Fairlight users, I spoke to Paul Saccone, Product Marketing Manager at Blackmagic Design: “The first thing we faced was getting the software across three platforms, as we were uniquely PC. I won’t go into details of what coding language we used but Fairlight’s small dedicated family of developers found themselves surrounded by Blackmagic’s big development team. Everyone was working collaboratively and basically online all day.” AudioTechnology: It didn’t matter that Fairlight was in Sydney, and Blackmagic had teams working in Melbourne and Singapore? BD: Wherever they might be. Blackmagic has tools that make this sort of collaboration possible.
Immediately a target was set; to get Fairlight inside Resolve by NAB. From the purchase it took a couple of weeks to assess Fairlight’s IP. Choosing the architecture we would use to code across platforms took another month. By then it was October and NAB was just six months away. It was like, ‘Holy crap! How are we going to do this!’ What the team did was fabulous. Obviously it wasn’t release-ready software, but the amount of Fairlight software that went into the NAB release was extremely impressive. AT: Blackmagic is calling it beta software and not pretending it’s the release version. I had to send a colleague in the US a copy of the Fairlight manual after he downloaded the first beta of Resolve and found very little info in its manual about the new Fairlight page. BD: You can see the priorities. There were some more fundamental issues to be solved. For instance, when you looked at Fairlight software you saw one piece of software, but it actually consisted of modules, some of which — like FMC, the mixer control module — can be traced all the way back to the pre-2000 FAME mixing system. Over time bits were added to each module and they had to communicate back to the other modules. Certain ‘inconveniences’ that had built up in Fairlight software had to be solved. Porting all of it to Resolve would have taken months, so it was rebuilt from the ground up. AT: Was it a chance to clear the decks and rewrite the software so it looks and functions like Fairlight but with fundamental low-level software changes? BD: Exactly! Previously it was impossible to do clip-based audio key framing within Fairlight. We’ve also had to introduce the concept of audio
WHY BLACKMAGIC BOUGHT FAIRLIGHT Paul Saccone (Product Marketing Manager at Blackmagic Design): We knew we wanted to do something with audio specifically in Resolve. Our mission with Resolve is that we are fundamentally changing the post production workflow by creating a tool that integrates all the disciplines in one environment so you can move seamlessly from edit to colour to audio without having to translate or export any files. It can be less of a linear process and more parallel where you can be doing audio, editing and colour at the same time. The Fairlight acquisition was a natural fit for the direction we wanted to go. We have gone from basic audio to probably the best audio in any video editing software ever. There’s a lot of things about the way Fairlight works that are going to surprise and hopefully delight video editors. These aren’t simply tools being bolted onto an NLE but a whole audio system. Users are doing it all themselves. Therefore putting all the tools in one place so they don’t have to learn three applications, or import/export stuff, is a natural evolution. For the higher end facilities they’re used to a linear work flow — edit first, colour, then sound. Building this tool that does it all for the indie film maker but also allows facilities to collaborate and work in a nonlinear way is really exciting. The response we are getting here in LA is off the charts.
FX rendering to Resolve. In the Fairlight world, user are able to perform 5.1 multi-bus mixing with over one hundred tracks, which would require 256GB of RAM if you used track-based and clipbased VST plug-ins without rendering. There’s also multiple mains mixes in Fairlight Version 5. Not only bus to mains, but sub busses to multiple mains outputs, which is all important to different versioning. AT: Will there still be a standalone Fairlight audio system or will existing Fairlight users have to learn to drive a video/audio finishing tool? BD: There will be a high level of faithfulness to existing Fairlight users. A Fairlight user is going to be able to load up Resolve and go for it. AT: For the first time ever it may be on a Mac or Linux computer? BD: Yeah. Last night I was working on a doco to upload to YouTube. The titles and audio levels were not quite right, so I copied the project from the PC I was using onto my Mac laptop, went home and kept working on it. Resolve archive files include used media so they are so easy to move around. AT: Are there any assurances you can give to Fairlight users worried old projects won’t open in Resolve? BD: There are already scripts and tools in the background that are going to be available for that. There has to be compatibility between existing DR2 projects and Resolve. As for legacy MFX3 projects? I don’t know. A third party could write translation software for that. AT: What about existing hardware compatibility and plans for future control and mix hardware? BD: You’re way down the line with future hardware. Those policies are a work in progress.
There are many generations of Fairlight hardware that work with version five software — older USB- and Ethernet-based models. With Resolve the line has to be drawn somewhere. I don’t see USB controllers being supported. However, USB to Ethernet upgrade kits exist for some USB controllers like the Xynergi panels. At NAB, Fairlight ethernet controllers were working with Resolve and the keys to switch to the Resolve pages were on those controllers. Also, iCan is now even more integrated with the software. [iCan is a control protocol allowing Fairlight controllers to drive other editing applications like Adobe Premiere - Ed]. AT: So if I have Resolve and Premiere on my machine, the existing Fairlight controllers can drive both NLEs via iCan? BD: Those are all still technical possibilities. How we will use that is a work in progress. The sky’s the limit, we’re very excited. PROBLEMS RESOLVED
Having the purchasing power and market muscle of Blackmagic behind Fairlight is a big plus for users of the DAW. As an existing Fairlight customer the future looks very good. Blackmagic brings hardware expertise, an astonishing speed of development, and a cost effectiveness that sees a full Fairlight system inside Resolve for US$299. As a Resolve user as well, it means for the first time I can do both high-end picture and sound post in the one software app without clumsily transfering between picture and audio post via unreliable interchange formats like AAF or OMF. It solves one of the weakest parts of the video post-production workflow. Exciting times for post people. AT 25
TUTORIAL
VIEW FROM THE BENCH
PART 1
Mic preamps might look straightforward, but Andy Szikla uncovers the many trade-offs of designing a ‘wire with gain’. Column: Andy Szikla
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In one of his books about lateral thinking, Edward de Bono talks about a bunch of designers sitting around a big table trying to re-invent the airplane. One guy wants to put in more seats so they can make more money per flight. Another wants less seats to make it easier for the peanut cart to get down the middle. One wants bigger fuel tanks so they can fly further. Another wants smaller fuel tanks so they can fit more baggage in. On it goes around the table as paradigm challenges paradigm, and conflicts are argued about and eventually resolved. Everyone there knows they all have the same goal, and that is to make a great airplane. However, they are also aware that at the end of their deliberations, if the thing won’t fly, they’ve all failed. Modern microphone amplifiers are to me very much like that, and a good design will be a harmonisation of different and sometimes contradictory concerns. A good microphone amplifier is not just a box that makes things louder. It has to have adjustment available to cater for a variety of audio levels at its input and output, reject external electromagnetic interference and unwanted signals, make its input moonlight as a DC power supply for microphones, exhibit low self noise, and offer high reliability. When you design a microphone amplifier, you are designing a whole system which is searching for its own equilibrium. Changing the parameters of any one section will often create an argument with its neighbour. I can’t think of another piece of audio equipment which appears so straightforward and simple, yet hides complex interactions between its various sections. ALL MICS SOUND DIFFERENT, SO WHICH ONE’S RIGHT?
A fully stocked recording studio will have a cornucopia of different microphones on hand. Why? Because they all sound different. So which one’s right? All of them. They all provide different flavours and colourations, which is the upbeat way of saying they all deliver sonic distortion. At its bare bones a microphone is a device which converts one form of energy — sound waves — into another form of energy — electricity. As different microphones represent different physical mechanisms for that conversion, it is understandable that results will vary between them. What we do in the studio is choose a mic which will sound best with the source we are recording; like when my Mrs Tech Bench chooses a deadly pair of shoes to complement her frock. The best studios also have the nicest sounding rooms. Do we care about sonic accuracy? Not really. Otherwise most records would be made in an anechoic chamber using a B&K reference mic, and Stephen Hawking would do the lead vocal. This brings up a much debated point. When designing or selecting a microphone amplifier for use, what importance should be placed on pristine reproduction, free of colouration, as one might expect from the design of a Hi-Fi amplifier? I think the answer to that question is, given the choice between the clean and the colourful, an individual
CARBON MICROPHONE
AMPLIFIED AUDIO
ELECTROMAGNET WEAK AUDIO
INPUT
OUTPUT
BATTERY
Electromechanical Carbon Amplifier The first electronic amplifiers, called ‘Repeaters’, were used in telephony from 1901. A weak signal at the end of a long wire energises the coil of the electromagnet, forcing a plunger to vary pressure against the carbon granules, changing their resistance. The battery current is varied by this resistance, producing an amplified output signal which is capable of driving several hundred kilometers of line.
user will choose what is right for their recording. There is nothing wrong with taking the sound of a microphone and preserving it, or changing it, or anything in between. It gives the design team a bit of room to move, but at the end of their deliberations, if the thing doesn’t sound good — they’ve all failed. ETCHED INTO HISTORY
The first recording equipment was not electronic, but entirely mechanical, and it was enough of an achievement just to have captured a sound and stored it. In 1877, Thomas Edison invented a gizmo he called the Phonograph. It worked a bit like when you stretch a string between two tin cans and you can hear your cousin. The Phonograph allowed you to yell into a funnel, and the sheer force of your personality would cause a needle to cut a groove into a revolving tinfoil- or wax-coated cylinder. You could play it back via the same needle, using a hand-crank to rotate the cylinder, and the funnel would then serve as a horn, to amplify the sound back to the listener. Initially there was no way of duplicating the cylinders, and if you wanted a hundred copies the artist would have to perform the work a hundred times. It took until 1901 to pull off making reproductions from a mould of the first cylinder, but it was a cumbersome and expensive process, and the recordings were only two minutes long anyway. Those difficulties aside, for fifty cents each, you could choose between marches, sentimental ballads, hymns, comic monologues, or a racist form of early ragtime known as coon songs. Meanwhile, Emil Berliner of Washington DC had come up with a flat disc version of Edison’s idea which he called the Gramophone, which possessed an important advantage. After the needle
scratched away the wax, you could acid-etch into the zinc plate underneath to easily create a reproduction master with at least three minute’s worth of material on it — more if you used a larger disc. This was technically the beginning of the record industry as we know it, but whether you were a banjo player or a symphony orchestra, you still had to shout down that funnel. Enter Alexander Graham Bell. Back in 1877 when Edison was inventing the Phonograph (also making him the world’s first audio engineer), Bell was inventing the Telephone — a revolution in communications which would change the world as dramatically as the internet did again a hundred years later. Like Gates and Zuckerberg in powdered wigs, Bell and Edison were riding at the helm of a new era. A year later Edison created the Carbon Microphone, which was arguably the first serious piece of pro audio kit ever. It consisted of carbon granules squished between two metal plates. Sound waves striking the plates varied the pressure on the granules, which changed the electrical resistance between them. Carbon microphones were a great leap forward. They were not high-fidelity devices, so their dominance in professional audio was limited, but their inherent simplicity and durability made them a standard component in telephones for the next hundred years. Edison’s carbon microphones were also ingeniously reconfigured as the first electronic amplifiers. Instead of using sound waves, a mechanical plunger was pressed against the carbon granules and, similar to a loudspeaker, varied its pressure magnetically in response to an audio source. The resulting signal that emerged on the wire could be as much as 100 times (40dB) louder than the original. These carbon amplifiers were used extensively in the new telephone networks, to AT 27
+20
Line Level Audio
0dBu
Instrument Direct Inject
–20 –40 –60
Mic Level Audio
–80 –100
TAKE A LONG LINE...
NOISE FLOOR
–130dBu
Noise Floor vs. Relative Input Levels This diagram shows typical level ranges you might expect to see from common sources at the input of an audio device. The ‘noise floor’ represents system and parasitic noise within the device. At a glance it is clear that line level signals are naturally less troubled by the noise floor than microphone signals, which may experience serious challenges. It is normal to try to design a microphone amplifier with the lowest noise floor possible, where -130dBu would be a resounding achievement, and -80dBu a bit of a failure.
boost diminishing signals in long-distance cables. In the years that followed, it was all about the telephone, and most of the important advances in electronic audio and amplification were made by people working for two subsidiaries of AT&T, the company founded by Bell. Western Electric and Bell Laboratories variously developed the vacuum tube, the first condenser microphone, some of the first valve amplifiers, and even invented the decibel (named after Bell) and negative feedback, but it took a while for these advances to find their way back into the record industry. The oldest known electronic recording is a document of the burial of the Unknown Soldier in Westminster Abbey, London, in 1920. The equipment used was a high output carbon microphone connected directly to a record cutting lathe. The result is noisy, fuzzy and indistinct, and like everything else, you can find a copy of it on YouTube. It also marks a starting point in the appearance of the first electronic recording systems, which were all ‘direct to disc’ and quickly saw the old mechanical technology wiped out of existence. By 1925, Western Electric’s ‘Westrex’ electrical recording system had established itself as the market leader, and did away with carbon microphones in favour of the sonically superior condenser mics. Class A valve amplifiers then boosted the weaker signal up to cutting lathe level, for the production of a wax master. These kinds of systems were commissioned into purpose-built studios all around the world, including the brand new Abbey Road facility in London (at that time, the largest recording studio in the world), and so the stage was set for the beginning of the musical recording era as we know it. AT 28
• Transformerless differential transistor input (Soundcraft 400B desk) • Transformerless instrumentation amp IC (Klark Teknik Midas XL200 desks) Interesting, at least to me, is that right from the get-go all of the above methods employed a ‘differential input’ as a way of suppressing unwanted external noise — and noise is the second most important matter a microphone amplifier must face.
Over the next 20 years the design of better sounding and higher fidelity microphones and amplifiers was to follow, with new researchers like EMI Hayes Laboratories setting many benchmarks, and manufacturers like Neumann producing classic equipment that is still hallowed today. Still, direct to disc recording remained the standard until 1948 when Ampex, 3M and Bing Crosby’s money finally developed magnetic recording tape into a viable alternative. NO GAIN NO PLAY’N
The principal and most tangible requirement of a microphone amplifier is to apply gain in the audio band of frequencies; lots of gain, and useful amounts tend to range between 40 and 70dB. In voltage terms, that is an increase in amplitude of up to 3000 times, which is quite a lot. The oldest systems accomplished this amplification using a combination of input transformer to class A valve circuitry, and that was the way it stayed until the invention of transistors. The transformers were wound in such a way that a small amount of gain was produced across the windings, and then output to the valve amp which would provide whatever additional gain was required. Various other schemes have been used over the years, and I have listed some common ones below, with examples of devices that have made use of those topologies. • Transformer to class A valve (Bill Putnam’s original UA100D and UA610 control panels) • Transformer to class A transistor (Neve 1073 preamp) • Transformer to op amp (Chilton CM series Broadcast desks)
Because a microphone cable is made from a long bit of wire it works pretty well as an antenna, and will receive all manner of emissions from radio broadcasts, digital TV, light switches sparking and the guy next door operating his hand drill. Then on top of all that, you have your mains AC voltage humming along at 50 or 60Hz (depending where you live). If all that junk infects our mic amp and gets amplified 3000 times as well, then we may as well call off the session and spend the night drinking. Differential inputs are our main defence against these troublesome unwanted signals, and a typical audio transformer can attenuate them by 100dB or more. In addition to our external woes, the components themselves produce noise, so every system has it, and you can hear it as hiss when you turn up the volume. For frequencies under 10Hz, semiconductors will produce a kind of crackle, like the slow grinding of metal on stone, as random charge carriers within the silicon combine or break apart. In a mic amp where the designers haven’t troubled themselves about noise, there can easily be something lurking in the background with an amplitude well into the hundreds of microvolts. We call that the ‘noise floor’ and what we want is for our audio signal to be as far above it as possible. That is one of the reasons why most of the downstream gear you are likely to use is designed to operate with signals of around one volt or so (professional line level signals of +4dBu correspond to an amplitude of 1.228 volts). Processing our wanted audio at line levels in the region of 80dB above the noise floor means it is far easier to keep it clear of all the unwanted crap, internal and external. A superior mic amp will have a very low noise floor, but in our inferior system a delicate ribbon microphone might only be in the clear by 20dB or less, and when we turn it up will sound like our artist is playing a tune in the back yard, while watering the garden. This helps explain why you should avoid plugging a line level output into mic level input. Every time you apply gain to a signal, you are also amplifying noise. If you have audio that is a happy distance from the noise floor, then it is not a good idea to turn it down, just so you can amplify it up again and bring the noise floor with it. It’s much wiser to connect line outputs to line inputs at the same level (see diagram 3). We no longer live in the era when it was good enough simply to achieve a result, and now when designers set out to develop a microphone amplifier that performs better than the average,
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noise management is a major subject that needs to be considered from the very first sketches. There is a subtle trade-off between noise and gain, and a more palpable one between noise and bandwidth. It can sometimes play out that several low-gain stages cascaded or added together might produce the same overall amplification with a lower noise figure than one single high-gain stage. But will that muck up the input impedance seen by the microphone? Or make our external noise rejection less effective? And how much frequency bandwidth do we want? Do we restrict it in order to lower noise, especially under 10Hz, or will that upset our many customers doing whale music? Or should we give them a switch that makes our amplifier operate down to DC? So finally, here we are at the big table where one argument starts to affect another, and hopefully the designer will find that process satisfying, and feel energised as they look deep into all the challenges for a unifying solution. Failing that, they might look deep into a beer. In the second part of this article, I will explain how differential inputs go about their important work of attenuating noise, introduce several other mic amp essentials, and talk about the design process, and the pursuit of what some people refer to as the ghost in the machine: that elusive quality of sounding ‘good’.
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AT 30
+20 0dBu
Line Level Output
Microphone Intput Line Level
Line Level
–20 84dB
Mic Level Out
–40 –60
40dB ATTENUATION
54dB
Mic Level In 40dB GAIN
–80 –100 –120dBu
NOISE FLOOR
NOISE FLOOR
Noise Gain from Mismatched Devices Passing a signal from a line level output to a mic level input can degrade system noise performance. On the left, line level audio with a Signal to Noise Ratio of 84dB is attenuated by 40dB to make it small enough to inject into a mic input, and the noise is attenuated too, which is good. Trouble is, when the signal enters the mic input on the right, it meets a new system noise floor. When amplified back to line level, the noise follows, and in this example we get 30dB more than we started with. Had we sent the signal without attenuation to a line input, our noise figure would have suffered minimal degradation.
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REGULARS
PC Audio There’s a new audio application update available. Should you install it immediately or wait a while? Column: Martin Walker
This afternoon I spent several hours updating my Internet browser after it claimed that a newer and better version was available. I’d been running the 32-bit version for years, and the 64-bit version was now beckoning. I duly followed the update instructions, and as I should have expected, a few minutes later had ended up with two installations — my previous 32-bit one and the new 64-bit version. Did I need both? Of course not. So I uninstalled the old 32-bit version, and then ran my shiny new 64-bit version for the first time. Of course, this streamlined new setup didn’t carry over all my carefully added browser extensions for tasks like blocking unwanted ads and saving extensive tab sessions. My tweaked screen layout had also been returned to its default state. I soon reinstalled my — thankfully, few — extensions, and fiddled with the screen display until I got my preferred arrangement back. Then it struck me. How many times have we all gone through similar procedures with music application updates, only to be left pondering whether all the effort was worth it? EMPEROR’S NEW CLOTHES
Now, it is wise to run the latest Internet browser version, and we should install major new versions of music software we buy to get the latest features, but blindly accepting free updates from every developer can be a time soak. I do always install any Cockos Reaper updates (my main DAW these days) that announce themselves when I launch the application because they’re always under 20MB, quick to install, and have never broken anything. I also tend to install new versions of most of my audio plug-ins if notified of their existence by email, largely because these invariably cure a few bugs and perhaps add a little more finesse or the odd new feature. However, I am increasingly wary of updating larger audio applications, and indeed more sophisticated plugins, at least without first checking what has changed via the release notes. For instance, it’s certainly not worth downloading and installing any free updates solely because the Mac code has been modified to comply with the latest Apple OS change, yet the PC code remains unchanged! AT 32
It’s rare that newer versions of an application are streamlined to run faster, and sometimes the bloat of extra features may even slow their performance. On various occasions I’ve had to revert to the previous version of an application because of unforeseen problems. After all, there are thousands of possible combinations of operating system, plug-in, extension, and so on. There’s nothing more frustrating than discovering one of your most used plug-ins grinds to a halt with a new app version. At which point you’re faced with rolling back to the previous version, or waiting for the plug-in developer to track down the issue, resolve it, beta test, and then release its own update! At the very least, if an update appears for one of your more sophisticated audio applications, wait a few days and scour the appropriate forums for relevant comments and feedback from other users before taking the plunge yourself. LATERAL THINKING
Sometimes you really need your wits about you and some lateral thinking to perform an update — here’s a recent example. I needed to update a sample-based plug-in to the latest version because a third party developer had commissioned me to design some presets for a forthcoming sound library. I spent an entire afternoon running in circles between two installer utilities from this one company. One of them would update the standalone version of the plug-in to the latest version but refused to update the VST plug-in versions I needed, while the other one only offered a slightly older version of them both that was some months out of date. After spending some time scouring on-line forums I finally found a workaround — ignore both utilities and manually download an even older version of the entire product from the company’s website, change its options during the process to force it to reinstall older 32-bit and 64-bit plug-ins over the top of my newish ones, and then run the latest install utility, which now ‘found’ the plug-ins and let me update both to their latest versions. What a rigmarole!
THE SPITTING IMAGE
Thankfully, examples as convoluted as the one above are rare, and also occur far more often in the opposite direction; clashes after a newer version has been installed, rather than problems during the installation process itself. However, my very best advice is to regularly use an ‘imaging’ utility to back up your entire Windows installation, so in the unlikely event an update (whether Windows-based or specifically audio) causes you any problems, you can roll back to your most recent image file. I’m not talking here about a backup utility, which generally only copies your data files and settings, but a disk image that not only holds copies of all those, but also the Windows installation itself, all your applications, and everything else on your Windows drive/partition. This is effectively like having a time machine, since once a disk image has been restored, your drive will revert to exactly the state it was in at the time the image was created. For this reason, I also do my best to keep all my personal data on a separate partition/drive to Windows, because then you won’t lose anything you’ve created in the meantime. System imaging was added to the high-end versions of Windows way back in Vista days, and from Windows 7 was even available on the Home Premium version. However, the Windows disk imager doesn’t have the best reputation for either the time it takes or the large size of the resulting files, so most musicians in the know tend to stick with the various third party alternatives that are free of charge for personal use, such as EASEUS Todo Backup (www.todo-backup. com/products/home/free-backup-software.htm) or Macrium Reflect Free (www.macrium.com/ reflectfree). They say data only truly exists when it’s stored in at least three different places, so store your image files on an external drive, DVDs, or at the very least on a different physical drive to the one used by Windows. Stay safe!
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STUDIO FOCUS:
SING SING SOUTH Sing Sing’s mothership studio on Gordon St, Melbourne, was a luxurious place decked with outboard gear, loads of instruments, a Neve console in one control room, and an SSL in another. With developers moving in on the storied spot, Sing Sing announced the studio’s secondary Chapel St location (Sing Sing South) would take its place as the studio’s spiritual home. A few hurdles accompanied the prospect of relocating the iconic recording hotspot. Moving a studio with this much gear is an elaborate process at the best of times, notwithstanding the fact that Sing Sing South offers significantly less room to put it all in. Sing Sing South isn’t a new space. The venue has etched its mark in Aussie recording history ever since the ’50s where it started out hosting radio plays. It fell under the Platinum Recording Studios name a few decades later to produce artists like Daryl Braithwaite, Kylie Minogue, and Crowded AT 34
House. When Sing Sing owner Kaj Dahlstrom purchased it in 1996, it initially acted as an overflow studio. “When we were full at Sing Sing in Gordon St we used to move work up there,” says Kaj. “Some people loved it so they just booked that place. We began moving out of Gordon St in January this year, so we started to do up Sing Sing South. We repainted everything and took all the great gear up. It opened on the 1st of June.” MOVING CONSOLES
So you’ve got to move an 88-channel SSL G-Series console from one studio to another. Where do you start? Thankfully, long-time Sing Sing engineer Adam Rhodes (pictured) is a genius and a half with pulling things apart. Adam successfully dismantled the console into three parts for transportation. Arriving at its new home on Chapel St, the SSL
Story: Preshan John
parts were carried upstairs to the control room designed mainly for mixing and mastering. Here, Adam pieced it back together as a 64-channel version of its original self. Quite a feat in itself, and probably not something you’d want to do everyday. Recording at Sing Sing South is a bit more intimate than its former location. The SSL tracking room isn’t huge but still houses a Yamaha G3 grand piano and six 16-channel headphone mixers for live recording. While the SSL studio is perfectly capable of recording bands (and often does), it’s more tailored for mixing, mastering, and overdubs. The Neve tracking space is larger and has three isolation spaces branching off it to accomodate vocalists, musicians, or other softer sources when recording live. Don’t sweat it if you forgot to bring your guitar because you’ll be spoilt for choice with the house options. Not just guitars, the Neve live room has a Yamaha C7 grand piano, vintage
Wurlitzer and Fender Rhodes electric pianos, and several bass and electric guitar amplifiers and cabinets. Not to mention comfortable couches everywhere you turn. Leisure activities are well taken care of with a games room, dining room, and another space with an upright piano and couches for working out parts between sessions. The boutique interior design is a nod to the studio’s rich history, with lots of cosy colours and wooden paneling making it a very friendly and comfortable place to cut a record. WANT IT MY WAY
Kaj says Sing Sing South’s USP is the studio’s ability to accommodate a variety of recording workflows. Pick an era and mimic its sound — because you won’t be limited by gear in this place. Everything you could want is at your disposal, from the 24-track two-inch Otari tape machine hooked
up to the vintage Neve BCM console with 16 channels of 1073 preamps, to a full Pro Tools HDX ecosystem. “If people want to mix in the ’70s they can go fully analogue if they want to. Or if they want to go ’80s they can still record analogue but do it on the SSL. But if they want to work in the box we’ve got all the plug-ins as well. The Pro Tools HDX gives over 200 input channels. Workflow-wise, we can cover anything from the ’50s right up to the current because we have the whole range.” It’s true. Sing Sing South is outboard central, and it’s a wonder there’s room for it all. Some of it was sold during the moving process but Kaj affirms the best vintage gear has made the transition without hope of retirement. After all, it’s what makes Sing Sing, sing.
SING SING EAST Sing Sing Recording Studios has recently partnered with Box Hill Institute which is in the process of building a new facility for its Audio and Composition program. Box Hill Institute has purchased a lot of gear from Sing Sing, including the SSL 9000 K Console, that will all be installed in a very spacious new tracking and mixing suite out at the Box Hill Campus. This will become Sing Sing East. Stay tuned for its opening in 2018.
www.singsing.com.au
AT 35
REVIEW
ASTON STARLIGHT
Small Diaphragm Condenser Starlight pinpoints your sound with laser accuracy and three different voicings. Review: Preshan John
NEED TO KNOW
With the Origin, it was a droppable headbasket. With Starlight, it’s a laser pointer. Trust Aston Microphones to add a peculiar touch to its first small diaphragm condenser. There’s not much about Starlight that holds to convention. Sure, it’s a pencil condenser, but whichever angle you turn it, there’s a refreshingly new element. Up top, there’s the black hump housing the laser. Down back, the tumbled steel chassis, that’s now become a trademark of Aston mics, is tapered like the shroud of a ray gun. At the very front is a sintered head basket. Basically,
PRICE Single: Expect to pay $549 Matched Pair: Expect to pay $1099
AT 36
CONTACT Link Audio: (03) 8373 4817 or info@linkaudio.com.au
loads of ball bearings tethered together to form an incredibly durable, yet porous material. AKG used to do it years ago, and Eric Valentine’s Undertone Audio console has a top sheet of the stuff, but it’s a rarity in microphone design these days. It’s a random configuration, so presumably any diffusion should be equally random, reducing head basket resonance. The Starlight isn’t all about lasers and raygun looks from the ’70s. It’s also got a three-position Voice selector on the body. It’s one of four switches that line the body — the others include
PROS 3 voicings in 1 mic Indestructible sintered head basket Laser is handy
CONS None
a two-stage high-pass filter, two-stage pad (-10 and -20dB), and the laser’s on/off switch. The mic feels solid and is fatter than an NT5 or KM184. It’s a fixed-cardioid transducer, and Aston hasn’t alluded to providing any swappable capsules in the future. Apparently the stock capsule was selected in the same method employed for Origin — a process of elimination by blind-testing multiple capsules with over 50 top producers. At 15dB EIN, Starlight’s self noise comes in a squeak high by today’s standards. Signal-to-noise
SUMMARY It’s got a laser, an indestructible head basket, and a ray gun look to it, but the coolest thing about Aston’s Starlight is its ability to sound like three different mics. You won’t have to worry about having the right mic for the job again.
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AT 37
Similar to the Origin, tracks recorded with Starlight don’t seem to require a heap of treatment… if there’s ever a mark of a good microphone, that would be it
ratio is 79dB A-weighted (94dB SPL) and you won’t blow its capsule until you hit an enormous 150dB SPL with the 20dB pad engaged. Starlight is available as a matched stereo pair, and if you purchase it as such, you get two nifty Rycote shockmounts and a stereo bar included. The plastic stereo bar doesn’t feel very resilient but it does the job fine. Little rulers on either side help you set up the perfect ORTF or NOS configuration. Standard dynamic mic-style clips are also in the packaging along with a helping of screw converters. Oddly, the Rycote shockmounts are threaded with a smaller 3/8-inch hole, which requires an uncommon thread adapter if your stands don’t already have the smaller screw at the tip of the boom. A minor inconvenience, but you might want to carry some adapters with you. TONE OF VOICE
The option to change the mic’s voicing is a really useful feature — it’s like getting three mics in one. Aston says this comes “courtesy of some very fancy front-end filtering between the capsule and PCB.” There are three options to choose from and the most obvious sonic change occurs in the high frequencies. Modern is the brightest (solid bump between 6-12kHz), Vintage the dullest, and Hybrid in between. Exact rolloff amounts are shown in the manual. I left it on Modern for the most flat and natural sound, but switching over to Vintage brings a ribbon mic vibe that works perfectly on hi-hats, guitar amps, brass, and other top-heavy instruments you need to tame. Just a heads up, switching voices while the mic is gained up will result in an almighty boom and potentially terminate client relationships if timed poorly. ACOUSTIC CALCULATIONS
Small diaphragm condensers excel on a variety of sources, not the least of which is acoustic guitar. Typically I look for detail, natural but not overpowering bottom end, firm mids, and un-hyped highs. Starlight instantly proved itself a beautiful match to my Martin acoustic. The amount of low end reproduced makes it sound a lot like a large diaphragm mic and you’ve got to keep its LFprone response from taking the reins by backing away a little. Setting the Voice switch to Modern opens up the top end without sounding zingy. The AT 38
full-bodied tone of the guitar remained intact in the context of a mix, only requiring some filtering to tidy up the lows. A pair of Starlights rocked on drum overheads, especially on the Vintage voicing for thicksounding cymbals and hats. Again, I can’t overstate how handy it is being able to change the sound of a mic three ways before it ever hits your DAW. Compared to a pair of ADK TT tube mics Starlight pulled the focus toward the meatier midrange, emphasising the thump of the snare and decay of the kick and toms. The ADKs sounded more open and lively, shifting attention to the sizzle of the hats and stick-y attack. Interestingly the smalldiaphragm Starlights picked up noticeably more low end than the large-diaphragm ADKs. At the end of the day, whether it’s a $100 mic or a $5000 mic, they’ll each probably have a role to play. Hearing when to use them is half the battle. Having three voice settings meant I spent less time worrying about which mic to put up, and more time dialling in the right sound. If you don’t have the right mic, you’re often reaching for corrective EQ to get rid of unwanted frequencies and put yourself in the ballpark. Most of the time I’d solo a Starlight track and do hardly anything at all. When it came to mixing, the tracks respond very nicely to processing, though most of the time just a little tweak was required to make it sit right. SHOOTING STRAIGHT
The laser deserves a mention. You know when you’re setting up drum overheads and you think you’re pointing them at the snare? Well, it’s surprisingly easier to miss than it seems — lasers don’t lie. How much does this affect the sound? It’s probably not enough to warrant a re-record, but it is another indictment on relying on your eyes when making music. While I like the concept of a laser pin-pointing the mic’s direction, the one thing it can’t judge is distance — which I feel would be far more useful. You can have the laser targeted at precisely the same spot on a pair of hi-hats, but a few centimetres forward or back will make a substantial change in sound. Perhaps a distance readout like a builder’s laser could feature on the next iteration. Whaddaya reckon Aston? If you believe strongly in mic position
repeatability, you can achieve it with stereo miking configurations like XY, ORTF, NOS, or wherever the mics are next to each other and angled. Nail the position, mark the spots on the instrument, and next time you’ll be able to get the exact same distance and angle by matching both laser pointers to the markings (provided you recreate the identical distance/angular relationships between the two mics). That’s getting deep into geek territory and I’m not sure everyone takes such a scientific approach to recording. STAR OF THE SHOW
Like the Aston Origin which AT reviewed last year, the Starlight has a certain character that’s hard to compare to another microphone. It’s a raw and authentic sound that errs on the side of truth rather than flattery. But it manages to sound polished at the same time. Similar to the Origin, tracks recorded with Starlight don’t seem to require a heap of treatment. And if there’s ever a mark of a good microphone, that would be it. I want to call Starlight a dark mic, but I won’t. It is less sparkly than other ‘industry standard’ small diaphragm condensers, and with a whole lot more low end response than you may expect. It sounds bigger than its size. Besides the sound, my absolute favourite feature is the switchable voicing. Three mics in one? I’ll take it. The laser has its place as well, especially for drum miking. It’s up for debate as to how much it’ll improve the sound of your mixes, though it will help mimic repeated mic positions with greater accuracy. Overall, the laser has very little to do with how much I like this mic. Price-wise, you’ll pay a bit more than double for a Starlight than a Rode NT5. While I love my pair of NT5s, they don’t have three switchable voicings, two HPF options, and a two-stage pad built in, not to mention a laser. Let’s shoot to the upper-class standard — a Neumann KM184 costs nearly twice a Starlight. Like the NT5, the Neumann doesn’t have built-in bells and whistles like the Aston does. So Starlight definitely gives you a lot for the price. Plus, of course, it delivers the kind of highquality results you’d expect from a professional studio microphone.
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www.sounddistribution.com.au AT 39
REVIEW
QSC K10.2
Powered Portable Speakers QSC’S K series has been a winner. Almost a decade later, version two still stands tall. Review: Mark Woods
NEED TO KNOW
First released in 2009, the original QSC K10 set the standard for portable powered speakers. It was already a crowded market but the K10 caught everybody’s ear by providing more coherent sound for the buck. The design was great; plenty of power, easy to use, the best carry bags, indestructible, well-priced, good looking, handles in the right places and multiple, flexible, mixable inputs… still, it was mainly the sound quality that got them in. I saw the enthusiasm spread through solo performers, folkies and duos, the acts that plug in acoustic instruments and sing for their
PRICE QSC K8.2: $1399; K10.2: $1499; K12.2: $1699
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CONTACT TAG: (02) 9519 0900 or info@tag.com.au
supper. They loved the accurate reproduction, the compact design and the surprisingly high volume, if required. The word spread fast and the reputation for hi-fidelity combined with ready power saw them move swiftly to the top of the class. Where they have remained. Nearly 10 years later the updated version has arrived. It’s hard to top a classic but time moves on and there has been some innovation in the last decade to consider. There are the changes you don’t see: both drivers are new and amp power is up to more than 2kW, with a hefty 1800W available for the 10-inch woofer and 225W for the 1.4-inch
PROS High sound quality Effective processing Big power Useful presets
CONS No phantom power
HF driver. Maximum quoted volume from the speaker has gone up 1dB to 130dB SPL maximum… and speaking of volume, let me take a moment to tell you what my pair of K10s survives every year. My wildest show is Chopped (dirt drags and bands to match) where they get strapped above the cabin of a tray truck with live bands playing on the back of the truck while it cruises the event… raucous bands with drums, amps and screaming vocals. I remain amazed at how much sound you can get out of a pair of 10inch speakers without killing them.
SUMMARY QSC has updated the K Series to keep it at the top of the portable powered speaker heap. It’s a big heap, but QSC has made these small boxes even more powerful, and useful, than ever.
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A NEW ANGLE
What you do see is an all-new design that retains the non-symmetrical QSC look and brings it forward a decade. It’s the same dimensions and weight as the K10 and still made from black impact-resistant ABS but the finish has changed from the distinctive near-gloss look of the K10 to a more subdued matte look. The concave steel grille over the front is a little squarer but the main difference is the cut off rear corner. It becomes the base when the speaker is laid on its side for stage monitor use. This corner has been re-worked and the side handle is now on the angled base section instead of the side of the cabinet. This makes it easier to grab if the speaker is face down on the floor but its wider to carry than the K10. The floor monitor angle has been increased a few degrees so it rakes up at a sharper angle. This is a good change as the speaker ends up closer to the performer. The chunky protective strips of rubber used to keep the side of the cabinet off the floor have been replaced by four non-slip rubber feet. I like the way the K10 strips wrapped a little way around the side of the cabinet for some extra protection but the K10.2 looks neater overall.
I remain amazed at how much sound you can get out of a pair of 10-inch speakers without killing them
SCREEN OFFS KILLSWITCH
The generously recessed rear control panel of the K10 has been made shallower and smaller as the higher power amp and new low-noise fan have seemingly pushed their way to the surface. The control panel has changed to reflect the times. Retained are the two combo XLR/jack inputs with XLR links and a single XLR mix out. Gone are the switches. The one I miss the least is the suddendeath mic/line switch on Input 1. The preset EQ options on two switches are handy on the K10, and I can find them in the dark, but gone, replaced by a screen and nudge buttons — I’ll get my glasses. To be fair, anything that makes the mic/line switch harder to accidentally engage is good and QSC recognises this by making you confirm your choice before enabling the switch to mic input mode. The screen also significantly increases the number of controls and options available. The original K10 had a few preset options on a couple of switches, the K10.2 has 11 factory presets. These tweak the speaker’s frequency response and dynamics processing to suit different physical environments or applications. There are settings for live sound, EDM, speech and monitors, both stage and studio. There are settings for using the speaker as a bass or acoustic instrument amp, and settings for handheld or wireless mics. The delay function is a great inclusion and works for me, I’m often running delay speakers from an analogue desk so it saves gear. Four-band EQ is available but is sensibly limited to cut only, and only up to 6dB at that. These speakers shouldn’t need EQ beyond the broad presets anyway and this minimises the effects of any crazy or accidental settings. Scenes can be saved to store user settings if multiple users are involved. The K10.2 is easy to use and navigating the screen is intuitive enough. Input 1 switches AT 42
between line and mic level, via the screen. Input 2 switches between line and Hi-Z, to become a handy high impedance input for instruments with pickups. Input 3 has changed from a pair of RCAs to a stereo 3.5mm minijack. Not sure about this, its sort of more modern but the common 3.5mm to RCA cables are readily available compared to 3.5mm male to 3.5mm male cables, and the reliability of minijack sockets worries me. Commendably there’s no Bluetooth, that really encourages low quality sound. There’s no 48V phantom power on the mic channel either. 48V on the line is nasty when the cable inevitably gets accidentally unplugged while its turned on but condenser vocal mics are common now... there’s a problem looking for a solution. BETTER, NOT BIGGER
The technology that gave the QSC K Series its superior quality sound is still there. QSC’s Directivity Matched Transition (DMT) synchronises the HF and LF driver’s coverage angle. Intrinsic Correction keeps the speakers tidy in demanding environments. With new drivers and more power I was interested to see how much difference there was in use, and yes, its new and improved. In an A/B comparison with the K10
the K10.2 produces the same type of warm, clear sound but it’s noticeably bigger. The low-end is fuller, the -6dB LF response is quoted as having improved from 60Hz to 56Hz and this extra depth is particularly noticeable on music playback. It’s louder overall and more resistant to feedback. The presets add another dimension, particularly if there’s no mixer or drive processing in the system or installation. Dance music on the dance setting sounds full and thumping. For live music shows with mixers the default setting is a flat start for FOH where EQ and compression will come from the desk or drive rack. For foldback, the stage monitor settings use sharp filtering to reduce un-wanted low-end buildup. Setting 1 reduces the boominess but retains strong lows for instruments. Setting 2 cuts more to focus on the vocals. In use I preferred the new, steeper cabinet monitor angle, it kept the performer closer to the speaker so the sound was more direct and used less space on stage. It can also be hung from M10 installation points and M5 yoke attachment points or pole-mounted, horizontally or angled down by 7.5 degrees. The K10.2 is a genuine multi-purpose speaker you could take anywhere. It looks neater and sounds better than the K10. Its more powerful and more flexible in use. Its reputation will be maintained.
AT 43
REVIEW
CHAMELEON LABS 7603 & 7603 XMOD
Microphone Preamps & EQ When Event Opal designer, Marcelo Vercelli, took over Chameleon Labs, he immediately started tinkering under the hood of its Neveinspired preamps. Review: Greg Walker
NEED TO KNOW
Since the early 2000s Chameleon Labs has carved out quite a niche for itself in the middle ground of the audio market. Its products are neither bargain basement nor pricey, and have always offered users great feature sets and a taste of what lies further up the audio food chain. With renowned engineer Marcelo Vercelli (Mackie, Event Opal, RCF) taking over the company in 2014, Chameleon Labs has undertaken several years of research and development resulting in the release of the 7603 and 7603 Xmod preamps. These are the first in a new generation of products that promise to see Chameleon Labs up the ante in its area of the
PRICE 7603: $1299 7603 XMod: $1999 CONTACT Chameleon Labs: www.chameleonlabs.com.au
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market. I was keen to take them for a drive in the studio and play them up against my vintage Neve 1272-based preamps to see how the new Chameleon Labs’ sounds stacked up. AMPING UP
Unpacking the two preamp boxes, it was immediately clear these are serious rack units with a fair heft to each of them and a high quality build. Dispensing with the silver faceplates and controls of the earlier product line, these units scale up the style with a matte black, anodised steel chassis, black aluminium faceplates, and tasteful blue, gold and silver milled aluminium
PROS Great design & feature set Plenty of Neve-style mojo in the sound Great EQ Switchable mic impedance & metering options Affordable but looks & feels pro
CONS May be too harmonically coloured for some tastes
knobs that are super smooth to the touch with centre-detents for easy resetting. Laser etched legending and small silver toggle switches round out the faceplate layout. The look and feel of these units is definitely pro and they sat in my racks looking very comfortable alongside units worth twice and three times their price. Speaking of racks, the width of the 7603 and 7603 Xmod chassis behind the rack ears is at the absolute maximum so if you’ve got home-made racks (as I do) you need to make sure they are wide enough to accommodate these units. I managed to just sneak the Chameleon Labs units into my rack with a bit of finagling.
SUMMARY Whether you get the in-house transformer or Carnhill-equipped version of the 7603, prepare for a bold mid range and plenty of ways to harmonically colour your sound. You’ll also have to ‘make do’ with great build quality and a huge feature set for a great price. The clone wars are heating up.
NUMBERS GAME
Like its predecessor, the silver 7602, the 7603 is very much a tribute to ‘revered classic preamps of the past’, which is to say the solid-state, discrete Neve 1073 preamp specifically. While the 7602 was also based on the 1073 design and layout, Chameleon Labs has raised the circuit design bar with several major design changes implemented in the 7603. Most notably there are new proprietary microphone and line-level input transformers that feature custom-made nickel laminations and bobbin assemblies. These ‘in-house’ manufactured transformers deliver high bandwidth and low distortion specs and are very musical in their implementation. The second main point of difference from previous designs is the new discrete audio input gain stage. All of the 7603’s operational amplifiers used in the unit are based on this new topology. Finally, Chameleon Labs utilises a new set of custom designed and manufactured inductors in the EQ circuits, further enhancing what was a popular feature of the earlier 7602 model. Overall, the 7603 offers a lot of features for a 1RU device. Apart from the input and output gain and EQ controls, there are nine toggle switches on the unit. On the left are three that select line/ microphone source, Hi-Z DI input and 300/1200Ω mic impedance (handy for optimising lower output ribbons, etc), while on the other side of the stepped input gain control another three switches select EQ in/bypass, phase and phantom power settings.
Over on the right hand side, the last three switches accompany a rather nifty circular VU meter. These switches select power on/off, input/output metering and metering dB range (normal and -20dB for gauging lower levels accurately). Amongst all these switches is a high pass filter rolling off 15dB/octave below 40Hz, 80Hz, 160Hz or 320Hz and three bands of inductor-based EQ that utilise the classic Neve EQ steps as well as adding more options in the mid and high frequency bands. Each band offers ±15dB of gain via continuously variable pots and the EQ bands as well as the filter can be individually bypassed to keep the signal clean when particular bands aren’t required. While the 7603 does offer a tremendous amount of control and fine-tuning of sounds, the front panel doesn’t feel at all crowded or confusing. Controls are easy to reach and a pleasure to use with the choice of smaller toggle switches doubtless keeping things manageable from an ergonomic point of view. TESTING TIMES
My first tests of the 7603 were driving a Soyuz SU-017 valve condenser as a drum room mic on a couple of sessions. My initial impressions were of a nicely balanced sound with plenty of character and I was happy with the straight preamp tone leaving the EQ section bypassed for this application. At higher input gain settings I was getting some nice soft-saturation flavour with a noticeable compression characteristic in the recorded
waveforms. The room mic sound worked extremely well and I ended up using a good percentage of this mixed with a dash of the close mics for the overall drum sound. Amplifying a ribbon mic for percussion duties and electric guitar tracking as well as a trusty SM57 dynamic microphone for close-miked acoustic and electric parts showed the 7603 could shine in a range of applications. Further uses of the 7603 cemented my impression of the 7603 as a versatile preamp with a nice character. Where I used the unit to overdub new parts onto existing recordings, the sounds blended in beautifully and I was really pleased with the tonal qualities the preamp imparted to stacked tracks. EQUAL BILLING
Patching the preamp into my console’s channel insert gave me a good opportunity to really explore the 7603’s EQ and it certainly didn’t disappoint. From the get-go I was favourably impressed by the EQ’s power and flexibility, with the overlap in the upper two bands complemented by the interplay of the high pass filter and low band giving the user some serious sound-shaping tools. On snare, the 7603 EQ allowed me to dial in a variety of valid snare tones; from tough and deep via generous doses of 60Hz or 110Hz to high and airy courtesy of boosts at the 12kHz or 16kHz steps. Vocals and guitars came to life vividly using the high pass filter and various combinations of midrange boost and cut while adding air and clarity to the top AT 45
end. The character of the EQ is quite Nevish in its way, and this can be emphasised via the saturation effects available when driving the input gain harder into the EQ circuits and backing off the output correspondingly. Even at more extreme settings the highs are nice and airy and the bass frequencies solid and convincing while midrange tonal shaping is very effective under both boosts and cuts. The overall sound of the 7603 is definitely thick in a Nevish kind of way while delivering detail and wallop in equal measure. Exactly how ‘Nevish’ I was about to find out when I lined up the 7603, the 7603 Xmod and my own vintage 1272-based preamp for a proper shootout. HIGH NOON AT THE NEVE CORRAL
Before we launch into the shootout, a word about the other two preamps involved. The 7603 Xmod is essentially the exact same preamp as the 7603 but with the new Chameleon Labs input and output transformers swapped out for more historically accurate and expensive-to-obtain Carnhill transformers — favourites with Neve cloners who want to get as close to the original Neve sound as possible with their recreations. All other controls and functions on the 7603 Xmod are identical to the basic 7603. As for the 1272-based preamp, it is not in point of fact a 1073, but uses an original Neve 1272 line amplifier module that goes a long way towards generating the 1073 sound with an additional custom gain stage to generate the desired output levels. The 1272 amplifiers were integral to the 1073 sound, and have been sought after for many years by the electronically inclined as building blocks for either bespoke Neve mixers or preamps. If you’ve heard the new Jen Cloher record on the radio of late you’re listening to this very preamp and its sister unit on Jen and Courtney Barnett’s electric guitars. These are my go-to preamps for thick creamy guitar sounds and larger than life snares and they never disappoint. But I digress… back to the shootout. I left the EQ sections of both the Chameleon Labs units bypassed to better assess each preamp’s character and started by returning to my original mono drum room mic setup. An identically played simple rock beat was the raw musical material here and the results were intriguing. The 7603 again sounded very nice with its smoothed out slightly compressed character to the fore, the 7603 Xmod seemed a little more aggressive in the mids but had a pretty similar overall flavour (which spoke well for the 7603’s inhouse transformers as a comparison), and the 1272 sounded, if anything, more polite and a little less compressed by comparison. More careful listening and balancing of gain structures revealed a certain amount of harmonic build-up around several key frequencies in both the CL preamps while the 1272 seemed more open and less inflected in this way. I liked the sound of all of them in different ways, with the CL preamps both delivering a slightly more mid-range focussed attitude and the 1272 a little more sweetness in the top end, a tad more low end ‘thunk’ and a slightly more even tonal balance. It certainly wasn’t chalk and cheese though, and all three gave valid and very musical images of the AT 46
simple but gutsy drum sounds on offer. The next stop was strummed acoustic guitar close miked with an SM57 where, again, all three preamps performed well. The 7603 Xmod seemed to lock onto the heart of the sound in a subtly superior manner to the stock 7603, while the 1272 again had a slightly sweeter top end and seemed to add a fraction less of its own ‘colour’ to the sound while also not compressing the transients quite as much as the other two. The 1272 delivered the most realistic sound and the 7603s added a pleasing colouration of their own. My conclusions were again quite similar when miking loud clean and distorted electrics with SM57s as close mics as well as using ribbons for amp room mics. All three preamps delivered extremely musical pictures of these sounds with the 7603s exhibiting a pleasing and slightly bolder midrange along with some useful dynamic smoothing, while the 1272 stayed a little more transparent both tonally and dynamically. PRECONCEIVED NEVE
It was interesting to compare these units and, to be honest, the results were in some ways opposite to what I was expecting. Subtle and super-musical harmonic saturation and larger than life low end are what Neve’s classic preamps are renowned for, and while my vintage 1272-based unit delivered these characteristics, they were overshadowed by the bolder midrange colouration of the 7603 and 7603 Xmod. The extra dynamic control these units delivered at higher input settings was quite a surprise, while the added harmonics in the midrange could be a plus or minus depending on the source material and the context of the other sounds around them in a mix. WASH UP TOSS UP
My overall impressions of the 7603 duo were of capable pro level preamps with plenty of character and great EQ at a very tasty price point. The difference between the stock transformers and the Carnhills is happily less than one might think, and for the money the stock 7603 in particular is a real winner. My one caveat regarding both these preamps is that their overt colouration may be a tad much for some applications and for large amounts of stacked tracks. When complemented by more transparent preamp options (such as those found in many digital interfaces) they will, no doubt, steal the show in many a mix. When used on key instruments such as snare, guitar and vocals, the 7603 and its Xmod sibling can be absolute winners for mojo-fuelled tracking as well as being extremely versatile equalisation tools come mix time. QUITE THE CHAMELEON
Chameleon Labs has upped the ante with the new 7603 range, adding custom transformers and inductors to an already feature-rich design complemented by pro build quality and audio performance. The Xmod model adds Carnhill transformer mojo and the EQ in both models is an absolute joy to use making the 7603 siblings great value.
A word from Marcelo Vercelli, CEO of Chameleon Labs, about redesigning a Neve 1073 circuit with modern touches: “We love analogue, it all starts there and some of the classic products we have come to appreciate over the years are truly amazing. One of those has been the Neve 1073. When I studied it years ago I felt that it was so very clever. When I contemplated just how we would confront a new design based on this existing one, I made a short list of things we wanted to work on in terms of developing something unique. “We first looked at the fact that nearly all of the classic audio gear we love has no star ground. We spent a lot of time focusing on ground plane design trying to achieve the lowest possible noise floor and maximum protection of the delicate mic signals arriving at the rear connector. We also had to develop a relatively sophisticated switch mode power supply with five voltage rails and ultra-low noise generation, all the while keeping high frequency noise away from the mic stage. “One of the super clever things about the 1073 is the way the input gain buffer uses a pair of stages that can be run in series to achieve the
70dB of gain they are specified to. Most of these circuits use a single-sided 24V power rail while we use a ±16V set of rails. More importantly we spent a lot of time developing a forgiving, low-noise analogue architecture using parts that are so much better than the ones available in the early ’70s and added a variable gain output stage that helps keep the unit quiet. The 1073 was based on the Fairchild BC184C, an expensive and worthy component in its day, but simply outclassed by the transistors available today. We have tried our discrete op amp design without the transformers and it sounds really amazing, no noise, lots of dynamics but lacking ‘character’. “The output transformers have a clear effect on harmonics and ‘character’. We spent quite a while winding input and output transformers with different types of laminations in search of
something we felt gave the product a unique character and found that the core and winding structure of the output transformer is where it all really happens. “At the end of the day we tried to keep things tidy and put some serious hours into making the product reliable and straightforward to manufacture.”
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REVIEW
AUDIO-TECHNICA ATM350A Instrument Condenser Microphone Audio-Technica upgrades its clip-on contender with more mounts than a GoPro.
NEED TO KNOW
Review: Mark Woods
Capturing the sound of acoustic instruments is one thing, amplifying them is another. Getting them loud and clear on the big stage is one of the more demanding gigs in live sound, especially if you want them to sound real. Some genres plug in with pickups and DIs but these rarely sound natural. Acoustic instruments are rich in fine detail with subtle dynamics and their beautiful, complex sound develops as it leaves the body of the instrument. Time has proven a high-quality condenser microphone is the best way to go. No problem in the recording studio where you can put the mic wherever you want and there’s no background sound but live shows are relatively uncontrolled environments and present special challenges for sensitive condenser mics, even high-quality ones. Gain-before-feedback is usually the first
consideration, especially for anything loud, followed by numerous physical considerations including size, weight and placement options. Audio-Technica has considerable expertise in designing mics for acoustic instruments. It all started in 1991 with the award-winning, Phil Ramone-influenced and promoted, AT4033a large-diaphragm condenser. Over the years this expertise has been used to develop a wide range of excellent instrument mics that have become staples of the acoustic genres. Each model has its own special thing but overall its condenser range is always, and fairly, described as offering ‘crisp and clear’ sound. I would add not overly bright or tizzy, with good low-mid warmth… and well made. Over the last couple of years I’ve noticed more players using the Audio-Technica ATM350 clip-ons as an alternate to the popular DPA
PRICE ATM350 Kits: from $599
CONS No ‘standard’ stand mount
CONTACT TAG: (02) 9519 0900 or info@tag.com.au
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PROS Sound quality Mounting options Good price
range. They look remarkably similar but are a fair bit cheaper and offer comparable sound quality. The mounting options were limited to an effective UniMount clip mount and a hook-andloop violin bridge mount. The ATM350a is the new version and features an updated capsule with impressive noise and SPL-handling specs, over-shadowed in some ways by the number and wizardry of the new mounting options. BUTTONED DOWN
The ATM350a system starts with the mic itself. It’s cute as a button, small but not tiny, and less than 15 grams. In use you don’t see it anyway, once you’ve wrestled it into the protective windshield it will probably live there. The windshield cleverly acts as both a pop filter and shock-mount. With the capsule inside, it mounts
SUMMARY Audio-Technica’s ATM350 was always a great clip-on mic, now the new version’s mounts let you get it into more places than ever. With two gooseneck lengths, and a collection of clamps, velcro, magnets, it’ll attach to stringed instruments, woodwinds, pianos and drums without getting in the way one bit.
The ATM350a is the new version and features an updated capsule with impressive noise and SPLhandling specs, overshadowed in some ways by the number and wizardry of the new mounting options
into loops on the top of the gooseneck and sits on a protective shelf that’s part of the gooseneck. Effective little clips connect the cable to the gooseneck for neatness and the whole assembly is strong and secure despite its light weight. The captive 4m cable feels soft and hangs nicely. On the other end of the cable is a neat detachable power supply unit with the HPF switch and an XLR socket for connecting to the outside world. There are two gooseneck lengths — five- and nine-inch — and a hexagonal bar at the bottom of each fits into one of the many mounts. This is when the fun begins. These mounts are a gadget lover’s delight; beautifully detailed and discreetly finished in matte black paint. The AT8491U universal clip-on mount will be the most popular. It’s the general-purpose option, designed to clip onto stringed instruments or the bell of many brass and woodwind instruments. The gooseneck is inserted into the mount at the chosen angle. It looks hexagonal at a glance so I was expecting six possible angles, but on closer inspection it’s an odd shape and you only get four, which is enough, anyway. The gooseneck is secured in the mount with a nicely knurled knob. The jaws of the mount have rubber teeth with a strong bite. Another knurled knob tightens the grip and can really lock the mount onto the instrument, so there’s no chance of it flying off in the middle of even the most frenzied performance. My old friends, My Old Friends the Chocolate Cake (hardly know them, just wanted to say it),
were playing at the Theatre Royal and as luck would have it the violin player Hope Csutoros was happy to try the ATM350a. We used the AT8468 hookand-loop violin mount first. Introduced as part of the previous model AT350 package it’s a Velcro wrap that straddles the strings between the bridge and the tail piece, the mic tucks inside a loop above the strings and points up at the instrument. Simple and quite discreet but it did seem to dampen the strings a little. Next we tried her normal technique, the clip-on mount attached to the violin’s chin-rest and pointing into the upper f-hole. The sound of the ATM350a was immediately impressive; natural, with an appealing warmth and closeness that brought a smile to Hope’s face and won instant praise from the rest of the band. The perfect reaction for something new. The gooseneck can be used to explore mic placements to find the best combination of tone and volume, and it doesn’t wander once in position. Hope is an energetic performer but no match for the AT8491U mount once it had been locked on. It didn’t fly off once. The frequency response has been tailored for live use with the deep low frequencies reduced to avoid rumble and limit the proximity effect. The high frequencies have been tamed a little to avoid the 10kHz-ish feedback that condensers sometimes induce, and the 2-4kHz region where live speakers like to take off has been slightly attenuated. Combine this with an accurate off-axis response and the result is excellent gain before feedback. The mic’s great… back to the mounts.
SCALING THE MOUNT
Next up was everyone’s favourite, the AT8491D drum mount, it attracted attention whenever I used it. More satisfying to play with than a fidgetspinner, and a lot more useful, its spring-loaded release attaches onto one of the drums tuning lugs. It’s secure, almost invisible and the mic will stay in the chosen position. I don’t think drummers like having things attached to their drums, it looks weird and those big rubber clips can change the sound. I prefer stands. This mount solves those problems. The clincher is the slip-on tension rod extender so you can still tune the drum with the
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mount in place. Ingenious. The first night I used the AT8491D was with some real rock bands. Peep Tempel were the headline act with not an acoustic instrument in sight, so I thought I’d try the ATM350a on the hi hat, since it’s a condenser… but it had no mount to attach it to a mic stand, an obvious omission when Audio-Technica has tried to cover everything else, so I put it on the floor tom. The drummer approved of the mount and was suitably impressed with the discreet placement as well as the way the drum could be tuned with the mount in place. Floor tom for a thrashy rock band is a long way from a subtle violin, and the ATM350a was the opposite of the tight dynamic mic I’d normally use on the floor tom for that sort of band. The floor tom sounded huge and the spill was good too… quite a lot of it, of course, but it blended in for a big, open drum sound. The mic had no trouble with the volume or the full low-end; all in a day’s work, as they say. The AT8491P piano mount is clever too, it may not be a unique concept but it’s an elegant solution to the problem of how to mount mics in a piano. Looking like the black queen in chess, the attractively shaped mount has a strong magnet hidden in its protective felt base and an integral shock-mount. The magnet sticks to the piano frame and secures the mount. The nine-inch goosenecks are recommended for this application. Even with the longer goosenecks the mounts are strong enough to hold the mics firmly in place and the look is super low-profile compared to two tall
boom stands. The AT8491W is a problem solver designed for woodwind instruments. Examples of mounting problems include the flare on the end of a clarinet being too wide for the basic UniMount jaws, or a flute that may be better captured by getting the mic nearer to the mouthpiece than the end. The AT8491W addresses these problems by attaching a rounded base to the body of the instrument with a hook-and-loop strap that passes through the base and wraps around the body of the instrument. The base has four soft rubber feet that touch the instrument and an integrated shock-absorber above the strap to reduce handling noise. The strap sticks to itself easily so there’s some skill required to get the fitting or adjustment process smooth, but once in place, it’s solid. The mic can now be placed to reach over the end of the flare as you would for brass instruments, or it can be placed to point back up the instrument, towards the keys, for a less aggressive sound, or aimed at the mouthpiece of a flute. Simple but well executed.
full and accurate view of the instrument for normal stage use but interchangeable elements are available with hyper-cardioid available for situations where players are very close to each other, and omnidirectional for super-accurate proximity-less applications. The mounts let the ATM350a do its best work. Single-instrument players will choose the mount that suits their instrument, multiinstrumentalists may choose several, while venues or production suppliers could get the whole set for maximum flexibility.
FLEXIBLE AS A GOOSENECK
The ATM350a is an excellent general-purpose instrument mic that will deliver the highest sound quality and robust amounts of gain before feedback in a live environment. Acoustic instruments and percussion have a low average level but can have surprisingly high peak transient levels. They won’t worry the ATM350a with its impressive 159dB maximum input level. The cardioid pattern gives a
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REVIEW
RODE VIDEOMIC PRO+ Camera-Top Microphone Review: Preshan John
NEED TO KNOW PROS Useful auto on/off feature Mechanically robust CONS Rear-facing output socket CONTACT Rode: www.rode.com
A good camera-top microphone to a filmmaker is like a trusty mini-mixer to a soundie. It doesn’t have to be the hottest featureloaded gadget to impress, it’s just got to work. The Rode VideoMic line has always had this utilitarian streak, but the new VideoMic Pro+ brings a little more bling. VideoMic Pro+ is a fair handful and chunkier than its predecessor. The Rycote Lyre shockmount is tucked in nice and close to the mic body, keeping it relatively compact when perched atop a camera. The 3.5mm output socket is still maddeningly mounted on the rear of the mic, meaning the rightangle cable pokes your forehead if you shoot video through your camera’s EVF. The cable length is a bit liberal for carrying audio the short distance from the mic to most DSLRs’ 3.5mm input. I kept it out of the way by looping it under the lens. Onboard sound-shaping options include a highpass filter with 75Hz and 150Hz cutoff points, and extra gain options of +20dB and -10dB. The +20dB jump introduces barely a whisper of noise and is perfect when recording ambience or soft-spoken piece-to-camera talent. Little lights indicate the mic’s settings and are easily visible in daylight. AT 52
I HAVE THE POWER
Whoever decided to add auto power on/power off functionality to VideoMic Pro+ deserves a medal. While out shooting I’m quite inclined to forget to switch the microphone on and off with the camera, only to later find the mic battery depleted after a few hours. Rode’s stroke of genius means the camera and microphone wake and sleep in perfect harmony, brushing those worries aside as a thing of the past. There’s literally no need to touch the mic’s power button. Still on the topic of power, VideoMic Pro+ can run off a pair of AA batteries or Rode’s own LB-1 rechargeable lithium-ion variety. What’s more, there’s a Micro USB port right on the unit for quick recharging. Thirdly, the battery door at the front side of the mic is now hinged to the body, so you’re not going to lose it. POINT & SHOOT
Naturally, the supercardioid pickup pattern of the VideoMic Pro+ is a long way from being a true shotgun mic, so don’t expect it to reject surrounding disturbances very much. The mic
picks up sound rather generously but anything front and centre feels closer, with more high and low end information captured. It’s a very natural and pleasant sound without the muffled character of many cheaper options. Null points are at 150° either side. Be aware of the rear lobe when shooting — it will hear things said behind the camera. This can be useful when vlogging interview-style or slating your shots. Handling noise is virtually non-existent, and you can easily press the filter or gain buttons mid-shoot without introducing a loud thump or click in the recording. The stock windmuff did a great job getting rid of rumble caused by a light breeze but if you’re shooting outdoors extensively a dead cat is recommended. LIKE MIC
With the VideoMic Pro+, Rode has stopped just short of completely reinventing the original VideoMic Pro. The auto-power feature is inspired, dual battery options invaluable, HF boost a lifesaver at times. Never underestimate how much a product can improve when you add a ‘+’ symbol. VideoMic Pro+ is a wise investment for all filmmakers and video hobbyists.
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