AudioTechnology THE MAGAZINE FOR SOUND ENGINEERS & RECORDING MUSICIANS
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REVIEWS: APPLE LOGIC PRO 9 UAD-2 SOLO/LAPTOP MC2 AUDIO E90 PROPELLERHEAD RECORD ALTA MODA UNICOMP DPA 4099 SERIES
MAKING THE CUT Behind the Score of Subdivision
DIGITAL & ANALOGUE One song, two takes, two mixes
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I’ve had a notion rattling around in my head for ages now, well several actually… but one in particular has been eating at me for a while.
conventional power rock kit, hitting it hard and expecting EQ, compression and/or mixing miracles to somehow turn it on its head later.
Often when I mix, I find myself carving bass drum EQ to make the instrument sound bigger. Actually it happens all the time; virtually every recorded kick I come across seems to have the same problem – a lack of genuine subharmonics, too much low midrange and not enough edgy mids. The result is the dreaded ‘cardboard box’ tone; you might even know it, the one that sounds like a knock at the door from an angry neighbour rather than a finely tuned drum. Sometimes that sound works well of course, but not that often.
The same lateral consideration of scale applies to anything you’re recording. If it’s a big, vociferous backing vocal, for instance, that you imagine way off in the distance, why not find an enormous warehouse somewhere and record it from 100 feet away? In that situation I always try and imagine that my mixing chair is literally in the space with the singer, and wherever I’m sitting in relation to them that suits my vision for that sound in the mix is where the mic should be placed. The same applies to a tight claustrophobic tone. If your imagination is telling you that all the elements of the production should sound like they’re crammed into a cupboard, do it! Record as many things in a cupboard or overdub booth as you can, even the drums. An instrument’s distance from the mic – whether it be 2cm or 200m – is critical to the sound you’re capturing, and judging that distance is one of the key aspects of good recording technique. Simply sticking mics close around all the instruments and worrying about scale during mixdown is a big mistake.
The internal debate this regularly generates inside my head while I’m mixing typically goes something like this: ‘Am I pushing this bass drum EQ too far or is the kick drum simply the wrong scale for the song?’ This topic is probably something I should tackle elsewhere in the magazine where there’s more room to manoeuvre but I just can’t seem to get it out of my head, so here goes… Audio production, and in particular, recorded music production, is a process that requires a keen imagination, part of which must involve the ability to perceive scale. From a production point of view this might mean anything from being able to imagine the placement of tiny voices far off in the background to weaving melodies with layers of instruments. Regardless of what elements are plucked from the universe of options, one of the ways in which instruments and sounds of a production come together coherently is through a keen appreciation of how they all fit proportionally together. Other issues like arrangement, volume, tone, pitch, bandwidth and tempo weave in and around this concept like spaghetti of course, but they’re not quite the same issue. Scale, like arrangement, is critical to the success of the production, and like arrangement, a failure to consider it independently of the others can lead to major problems. Without scale in mind, maintaining a vision of the final outcome becomes exceedingly difficult. The trick is to be mindful of choosing the right instruments and trying to imagine how they fit together in the production scene long before the mics come out of the drawer. If you’re contemplating the scale of the drums, for instance, then the size of the shells, the dimensions of the recording room and how hard the drums are hit are all critically important things to consider. If you imagine the final mix dominated by lightly struck, tiny-shelled drums then it’s vital to set-up that scene physically in the room. It’s no good setting up a AT 2
Whenever you have a clear image of what something should sound like, act on that vision with confidence. Hearing the recording as you imagined it will immediately satisfy your brain’s hunger to ‘see’ that idea realised. Not only that, but it will make additional arrangements easier to conceive and easier to perform because the scene is being set as you go. Once you get into the habit of capturing sounds as you imagine them to be, your recordings will improve and your reliance on reverbs, delays and other ‘scale modifiers’ will lessen. What’s more, recording with scale squarely in mind will also give you a much better understanding of the role effects like reverb and delay ultimately play. I’ve been EQ’ing, delaying, doubling and reverberating things to modify their scale for years now, yet much of this ‘scale management’ is misunderstood to be an issue specific to mixdown. In reality it’s arguably more fundamentally tied to the tracking phase of a production. This is one of the reasons why mixing sometimes takes so long and can become so arduous. When 50 close-miked instruments are suddenly ‘rehydrated’ back into a space, everyone involved in the process is suddenly awakened to issues of scale, depth and distance. And, that’s when things get tricky. As we all know, people often ‘see’ mixes differently, especially when they’re denied this vision during the tracking phase. That’s why it’s a good idea not to leave this ‘awakening’ ‘til mixdown – the longer you leave it, the more trouble you can find yourself in.
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All material in this magazine is copyright © 2009 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. 18/09/09.
CONTENTS 70 – Full magazine out now
FEATURES 26 MAKING THE CUT
Brendan Gallagher lands his first film-scoring gig and takes AT along for the ride. 32 MUSIC FROM THE CELLAR
Australian country music’s dynamic production duo, Rod and Jeff McCormack, invite us into the Music Cellar where many hits are penned, tracked, mixed and mastered. 38 MOOT OR MUTE?
Chris Vallejo sets up a band and tracks them to two-inch tape, and then repeats the performance on digital multitrack. It’s a mix competition with a twist… you be the judge.
REGULARS 10 YOUR WORD
Readers’ letters. 16 NEWS
News and new product information, including a sneek peek at the new d&b T Series PA. 50 HOME GROWN
Jim Moynihan, AKA ‘Spoonbill’, shows us how his unique production style comes together. 54 WHAT’S ON
62 P C & MAC AUDIO
This issue, Martin Walker investigates Intel’s Core i7 and i5 processors while Brad heads for the snow and discovers bigfoot: 64-bit kernels. 96 HOW YOU GET THAT CLASSIC TONE
Rick tries to set the record straight about tubes and their so-called ‘warmth’.
TUTORIALS 46 I’M ALL EARS
Switching to in-ear monitoring is still the great unknown for many, but it’s a decision more and more people are making. So how do you do it? 58 STAV’S WORD
Stav mixes back to front with the assistance of real-life faders. 60 ON THE BENCH
This issue Rob Squire urges us to investigate some DIY kits and learn a bit about what lies beneath the products we all love.
REVIEWS 66 IN BRIEF
BBE D82 Sonic Maximizer, EMA A1 Generics, Beyerdynamic MCE55 68 APPLE LOGIC STUDIO 2
72 DPA 4099 SERIES
74 MILAB BDM-01
Bass Drum Condenser
76 ALTA MODA UNICOMP
78 PROPELLERHEAD RECORD
82 MC2 AUDIO E90
84 GILES AUDIO VD-1
86 MELLOWMUSE ATA
88 MOTU VOLTA
MIDI-to-CV Conversion Plug-in 92 BLAK HOLL
94 UAD-2 SOLO LAPTOP & QUAD
Plug-in Emulation Hardware
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A UAD-2 Solo/Laptop! AT 3
Making the cut -scoring Brendan Gallagher lands his first film gig and takes us along for the ride. Text: Brendan Gallagher
I’ve been trying to get a feature film score for 20 years. I can remember driving a taxi and having cassettes of my music with me just in case I met an industry player. I had actor Kate Fitzpatrick bailed up and bemused in my cab in the late ’80s listening to my tapes outside her house at the end of a fare. She was very sweet about it. Always carry your product with you – you never know who you may meet in the lift. In the last 10 years I’ve done soundtracks for TV docos, short films (Warwick Thornton’s Mimi, for example); I even played bouzouki on a pizza ad once. I’m a song guy but occasionally I stray into cinematic territory with my arrangements – strings, horns, dynamics, space… A year ago my publisher Universal Music put me in touch with music supervisor Andrew Kotatko [see ‘The Super’ sidebar] who was casting around for candidates to compose the music for a new Australian feature, Subdivision. He’s a fan of Jimmy Little’s Messenger – an album I produced that has become a calling-card for me ever since – and he thought I would suit the job. STRIKING A DEAL I had a meeting in August 2008 with the Director, Sue Brooks, and Editor, Jane Moran, in Melbourne and had a look at the early cut of the film. These meetings are something of a beauty contest, to see if there’s any chemistry between the various parties. The role of composer is to
carry or augment the emotional tone of the film so you have to be on the ‘same page’ as the director, as the Americans say. I got the job, great news. The producer rings me to go over the budget. ‘Cut your coat according to the cloth’ the old saying goes, so I make a guesstimate about what I can do with what dollars are available and still make a living. That’s a big part of what producing is about. Start with the result, work your way backwards from there and you get an idea of what’s possible. SHAR ING A LAN GUAG E Then it was back down to Melbourne from Sydney to go through the nuts and bolts of the score. Which parts of the film need music, what kind of music, tempo, texture? This is when you need to be able to get on the same wavelength as the director and the editor.
Mostly, I’m a singer/songwriter who produces his own recordings. I get an idea, develop a song, flesh it out with a beat and dynamics and come up with a track. Not so with film. I have to take my cue from the film’s language. In my early meetings with Director Sue Brooks I told her that I work instinctively, that I wasn’t particularly cerebral about ideas of theme and motif. I would somehow absorb the idea of the movie, osmosis like, and arrive at some kind of coherent palette of sounds for the score.
After a few spins through the film, which has a working class, small community vibe – everyone works together, plays footy together (it’s set in Hervey Bay, Queensland) – I get some ideas for establishing cues. Handclaps become a prominent feature. The main character fancies himself as bit of a ladies’ man (like matinee
idol Clarke Gable… in a regional Queensland kind of way), which lends itself to ideas of post-war, small combo, swing music – vibes, brushes, double bass, chromatic harmonicas and whistling. This is my in-joke, this is what informs me, the way I can keep having fun with the film when things get bogged down in messy, repetitive detail. 20 years ago I interviewed legendary Australian composer Peter Sculthorpe for a radio doco about film music. His score for the ABC TV feature Essington, about early European settlers in the Northern Territory, was mainly a piano piece. The whitefella themes were played on the white keys, the blackfella themes on the black keys. Only he knew, but it was his in-joke, his subliminal narrative which informed his enthusiasm for the project. I have tried to employ the same idea in my work ever since.
SETTING UP CAMP So far that’s all right side of the brain stuff. What about the tech side? As most AT readers would know, the 21st century is the province of the novus homo – self-made person. You write it, you play it, you record it. I’ve been doing it for 30 years and it’s second nature. It’s obligatory in this lean and mean contemporary music scene. You have to be able to disassociate the artistic and the technical faculties of the brain, literally. The key is decision making. As we read in a recent interview with George Massenburg:
“choice is the enemy of commitment”. Too true. I’m fortunate to have some good working environments. I have a studio cum edit suite in the basement of my block of flats. There’s an iMac G5 in there with some chunky plug-ins, a Digi 002 with a Focusrite interface, a TLA stereo preamp/compressor and a TAC/Amek 24-channel console. I’ve also got a few mics – a Shure 58, 57, an AKG C-414 – but it’s mainly a ‘hothouse’ space where I work through ideas and edit stuff, often times tracked in other more expensive audio environments. The other place is at my mate’s studio Gareth Stuckey’s (aka Gigpiglet) – GPHQ in Redfern. I helped Piggly build the studio a few years ago and have a few toys that live there – a Leslie speaker, Gretsch drum kit and percussion, some guitars, some organs, my cherished Quad Eight preamps, a pair of UA 6176s, a Studer B67 1/4inch stereo tape recorder etc. We have a good working relationship and he looks after me when I need some studio time. GPHQ has evolved into a great workspace: there’s a Mac eight-core with ProTools HD, 32 channels of Lynx converters, a DDA DMR12 (an ex-Alberts, 56 frame console) and some great front end like four Peach Electronics 192 valve preamps, a pair of Distressors, a Rupert Neve CIB stereo mic pre/compressor, a Universal Audio 6110 preamp/compressor, a TL Audio Ivory 5013 stereo valve EQ, two dbx 160s, a Drawmer 1960 stereo preamp/compressor and Meyer HD-1 monitors.
A disclaimer: I am not an engineer, I learnt to record the same way I learnt about the piano – I opened the lid. Engineers are people like my mentors Phil Punch and Don Bartley who know stuff about voltage and impedance. I don’t know how the toilet works.
MISSION POSSIBLE So I’m set with working spaces, now to the film tech minutiae. I was fortunate enough to speak with composer/sound designer Rob McKenzie who works out of Soundfirm’s post production suite in Melbourne before I embarked on the project. He’s worked on some big numbers like John Wu’s Mission Impossible II (say no more) and dispenses me some wise and informed advice: record at 24-bit/48k, stems by bandwidth, no limiting, and then some – a valuable 30 minutes of ‘face time’.
The assistant editor is responsible for supplying me with the QuickTime copies of the six-reel ‘fine cut’, approximately 15 minutes each. I import the QuickTimes into ProTools and spot them into the corresponding timecode. Each cue – and initially there are 21 with some variations – is saved as a separate session. This is very important… modern recording is all about archiving. File path is everything. QuickTimes are a great and easy-to-use platform for film composition. I can remember using stereo VCR tapes with timecode on one side and vision on the other with a Tascam MX2424 in chase mode – not always reliable. COMPOSE YOURSELF The composition comes first and I usually like to keep it bone simple. I look at the scene I need to write to and establish a tempo. I’ll just sit there and tap my foot; I intuit the scene… all very herbal. As much as I loathe hippiedom (did somebody say djembe?) it’s an organic and reliable approach. Then I formalise the tempo in Reason with a simple drum kit pattern – just a kick and snare with no hats or cymbals because they have a bad habit of leaking out of headphones when tracking later on.
draughts of music so protestations of ‘but it’s just a demo’ will ultimately be a waste of your breath and time [see the ‘Temp Work’ sidebar]. So any ‘big picture’ items in sonic terms tend to get noticed and fetishised; the supporting elements, like pads, percussion etc not so much, those bits you can change without too much drama. If later you change any big picture items, even slightly, there is a chance you could put the grownups off balance. For example, there’s a scene where the leading man Jack, a builder, is officiating the handover of his first house to the new owners. There’s a long crane shot of him entering the house with champagne and glasses; he’s feeling pretty high on himself, the aforementioned matinee idol moment. I wrote a swinging, combo piece in the style of harmonica legend Larry Adler – double bass, brushes and arch top guitar. For the melody I did my best Morricone/Sergio Leone whistle, as a guide for the harmonica player (chromatic genius Christian Marsh). The producers, the director, the Fed Ex guy, everyone just loved the whistle, and despite doing a couple of eye-
Once I have a tempo, I can work on the textural stuff – establish a patch of chords on a guitar, sometimes a keyboard, or a melody, it depends on the scene. This is the mysterious aspect of composition, something that other writers would attest to, where you start to channel ideas and form them into a piece. I generally keep it simple, but any ‘moving air’ elements I may use – i.e. acoustic instruments – I record properly for the simple reason that I don’t have to do it again later. Remember, producing is decision making. NOT JUST A DEMO Directors and editors, producers too, have a tendency to become attached to original
The input rig is a messy old affair – but effective.
The producers, the director, the Fed Ex guy, everyone just loved the whistle, and despite doing a couple of eye-wateringly beautiful harmonica takes, one low, another an octave above, the original whistle stayed.
wateringly beautiful harmonica takes, one low, another an octave above, the original whistle stayed. Some days you get the bear, some days the bear gets you. BIG REAL ESTATE ITEMS The first set of cues I attempt are some of the more rhythmic, complex ones which will have a lot of elements to track downstream – drums, bass, horns, percussion, electric guitars etc. The moody, simple stuff with pads and simple instrumentation I leave for later. For now I’m concerned with the big ‘real estate’ items which will absorb a lot of my time.
When I’ve got some cues I’m happy with I email them as MP3s to the assistant editor with all the relevant timecode info so he can sync them into the soundtrack for the editor. I should point out at this stage that I’m based in Sydney, the editor and director in Melbourne and the producers in Brisbane so there is room for error, which occasionally happens. A few phone calls, many emails and a few trips to Melbourne and Brisbane to meet with the editor and director ensue to finalise what works and what needs changing. So I revisit some cues and try some other ideas. About 60% of the cues pass muster first time, others take up to four or five drafts before everybody is happy. SHAKY SCHEDULE My original delivery date is for late October 2008 but by late September this is looking shaky. There are problems with the cut and the opening titles are being re-shot, so the date is being pushed into early November. I have to keep my cues fairly elastic at this stage and not commit too much real estate. If a sequence gets shorter or longer, then the music cue changes. There are
Actually I turn it to quite an advantage. Early in my conversations with the director she mentions that if I have any songs that might work as ‘source’ music then by all means present them. This, forgive such an egregious pun, is music to my ears. Anyone who has been in the biz any stretch of time will tell you that back catalogue is everything.
all sorts of programs to stretch or shrink music but more often than not they become ‘Slow Tools’ moments – in other words, it’s often faster and more satisfying to just redo it. By this stage I’ve composed most of the requisite music and everybody’s happy with the demos. But I don’t have a final cut of the film so I can’t commit. This creates logistical problems. I have an east coast tour looming with Karma County; GPHQ has been locked out for six weeks to track and mix; I’ve just turned down a national tour playing guitar with cabaret sensation Camille O’Sullivan (what was I thinking!); and a series of songwriting workshops at Australian Institute of Music are going begging because I thought I’d be in the thick of tracking. There are at least 10 other musicians I’m trying to co-ordinate for sessions, some of them recording together, all of them with demanding schedules like me. The edit drags on… there are problems. Eventually I call the producer and lean on him for more money, (a) to pay a cancellation fee for the studio and (b) to compensate me – I’m working way beyond the agreed schedule and I’ve lost potential income. He graciously agrees. COURT ING ADVAN TAGE Ultimately, after three different re-schedules, the film is locked off in late December. In the interim, I track some elements of the cues that are definitely not going to change and that I can do on my own – guitars, some bass, keys, percussion etc. There’s still a lot of downtime to kill so I do a deal with the Gigpiglet and in the interim I mix a bunch of tracks for a solo album I’ve been working on, turning disadvantage into advantage, another crucial element of producing.
The downside of being an artist without a record company to bankroll your recordings is that you have to pay for it all yourself – get a day job, max out a credit card, borrow from your mum – whatever it takes to pay for mixing, mastering, artwork, marketing etc. The upside is that you own the recording copyright, this is a very good feeling, just ask Prince or George Michael who don’t own the masters of their early and most sought after recorded music, some big major record companies do. I own or share ownership in the sound recording copyrights of at least a dozen albums, most of which I’ve written or co-written. So if someone wants to use a recording of one of my songs they have to talk to me and my publisher about licensing it. Ergo, I get paid – it’s the gift that keeps on giving.
So I have a few, nay, many, source music ideas that I throw at the director and editor. One is The Millionaire$’ version of the Doris Day number Sentimental Journey, which plays over a punch-up scene in a footy match. It stays. Another is Karma County’s Lifesaver’s Love Song over a beach love scene between the two main characters. It doesn’t. A new song Try, Try Again, which I have rough mixed, gets a big thumbs-up in a pivotal scene where the lovers have a falling out. Luckily for me the film’s main financier loves the song too and it also gets used on the closing credits.
Brendan on bass
TEMP WORK ‘Temp’, or temporary, music very often becomes permanent music which is why the smartest thing any aspiring singer/songwriter/ composer can do is befriend film editors, or film makers in general – whether they be established or still at film school – and load them up on a regular basis with your latest work. In a fractured music delivery environment third-party usage of your work is a very important avenue for anyone serious about making headway in the music biz; (a) you get paid up front, (b) people who would otherwise not hear your work, because radio is not playing you, are exposed to your unheralded genius, (c) it’s cool when your song comes belting out of the speakers in a big cinema and, did I mention? (a) you get paid.
STAYING FLEXIBLE The song is too long for the lovers’ tiff scene so I do an edited version and the film editor re-cuts the scene a little to make it work – a bit of flexibility between composer and editor, nice. I’m running a little behind so I hand the two versions over to Tim Whitten to mix. I love Tim’s work and delegating is good practice, so I’m told. Meanwhile the film requires a song over the final scene, one which they want me to write. Not only that but they want it to be along the lines of Feist and Sally Seltzman’s hit song 1,2,3,4 which has been the temp music since the early cut but for various reasons can’t be used. Hmm, not so easy. It’s like asking a playwrite: “Hey, we need something like, you know, Othello.”
The director rings me up and asks me to keep in mind the concept of ‘inclusion’ when I’m writing. I come up with a song called So Said Muhammad Ali (it’s a long story). They like it – right sentiment, same tempo, same length as Feist’s tune. Apparently the film has a tie-in, however tenuous, with Sony and they would like to use one of their people. I look through Sony’s roster and the only artist that rings my bell is Bic Runga. I love Bic Runga, I would wash Bic Runga’s car and clean her house for a year if she would come and do a song with me. But it turns out not to be an option. Bugger… I broach an original concept with the producers – if we can’t get Sony to play along, why don’t we just get someone else who is, you know, just good. Works for me. So I approach Abbie Cardwell who I’ve always wanted to work with. She sings, she plays banjo and harmonica, she has a lot of X factor, and I know her. She helps me finish writing the song and, via MP3s and email (I in Sydney, she in Brissie), we do a demo that everybody likes. ROLE CREDITS All this, the song, the source music, the cues,
Jim Ellio t on the kit
Abbie Cardwell plucks
the locked edit is pretty much finalised. It’s five days ’til Xmas, nothing’s going to happen, it’s silly season, so I book all the players I need for sessions in January. Between Xmas and New Year I tidy up the cues for the locked edit. The opening credits are one of the first big jobs I tackle. It’s an important cue. It’s nearly three minutes long and has to be dynamic and rockin’. After three composition attempts I have the right cue. I lay down a drum click in Reason and track an acoustic guitar as a bed. I record acoustic guitars a lot and usually like this: for my standby acoustic, a Gilet J180, I use an Oktava MK 012 pencil mic placed about 30–40cm out from where the body meets the neck and facing the sound hole. I generally roll off 3 to 5dB around 200Hz, where a big body guitar has a lot of boof, and compress it a little, around 3dB with 4:1 ratio. I’ll also record the guitar through a BBE Acoustimax active DI – I use it on everything: basses, electric guitars… The late, great Jackie Orszaczky reckoned he put vocals through it.
The front end of the mic and DI is a pair of Universal Audio 6176 preamps, bought when they first came out, love ’em. I can pull a sound real fast and the auxiliary inputs on the front are a bonus. For 90% of this project I go straight into ’Tools through a box of effects I have set up on an old hospital cantilevering platform ($5 at a garage sale) suspended above the outboard rack and into the 6176s. It keeps it tidy, I can stand there (I work alone 90% of the time) with a guitar or bass strapped on and twiddle knobs ’til I’m happy and can push it back out of the way when I don’t need it (see pic). In the rack there’s the Acoustimax, a Rich Class-A distortion pedal,
a Holy Grail reverb, Boss orange distortion and blue chorus, Boss eight-band EQ, Roland Space Echo pedal and a DOD Octaver. RE-MIKE-ABLE One of the most useful bits of kit I’ve bought recently is the Broadcast Passive Link. It’s great for re-miking recorded tracks, mostly electric guitars, sometimes keys. When you’re working on your own it takes five times as long to get a good ambient recording, for example, of an electric guitar. If I already have a good DI recording I can run a line out of ’Tools into the Passive Link, which delivers line level input into an amplifier – two channels if I need it. It also has a volume and tone control which makes it just that little bit sweeter to drive an amp just right.
From there, I can set up a few different amps, a couple of mics, highlight the audio file in ’Tools and loop it while I tune an amp, set the input on a preamp etc and re-record it. On the opening credits I recorded a particular electric guitar drone that I like to use. You tune the guitar to an open chord, in this case an A7sus4, and place a slide over the twelfth fret and pick the strings between the slide and the nut, not between the slide and the bridge. While moving the slide back and forth, the overtones on one side play flat and sharp on the other – it choruses. It’s an old slide player’s party trick but very effective and ghostly sounding. I re-miked this through an old Goldentone vibrato amp out in the GPHQ kitchen area – five different takes, each successive take with the vibrato speed advanced 20%, from ‘surging’ to ‘space wobble’. This became a kind of wind chime motif, very Queensland seaside, which appears several times in the film. In one scene where there is a bit of drunken biffo action it’s played at half speed – gives it a syrupy, slowmo humid feeling. SESSION TIME It’s the second week of January and I start doing sessions. I hire an assistant, Annika Unsen, who I met doing production workshops at AIM –
“ I like to spread a mix
out on a desk, give it some real estate, get my hands on the knobs, walk around the room eating a sandwich instead of looking at a screen.
Over the next week I track conga and vibes, horns, piano, organ, cello, vocals, banjo and harmonica. I go around to (keyboards wizard) Stu Hunter’s flat to record an organ pass for the hero song on his Hammond B3 but the power amp blows up when he turns it on. Bummer, I really wanted to hear that puppy. So we re-schedule to do it back at GPHQ with a Nord through my Leslie.
THE ‘SUPER’ A music supervisor coordinates all aspects of the required music. He finds and interviews appropriate composers for the ‘underscore’ – the music composed specifically for the soundtrack. They also assemble and recommend a repertoire of possible ‘source’ music – already existing recordings, for certain scenes e.g. Stealers Wheel’s <<Stuck In The Middle With You>> from Mr Blonde’s torture scene in <<Reservoir Dogs>>. The music supervisor deals with the relevant publishers who administer the song composer’s rights to clear film usage and establish an acceptable synchronisation fee. The same clearance and fee process has to be done with the owner of the sound recording as well, usually a record company. It would be the exception that the same entity would control both rights.
she is very cool and a great engineer. This allows me to concentrate on the music a little more. For several cues, including the opening credits and the hero song, So Said Muhammad Ali, I use drummer Jim Elliott (Cruel Sea) and Michael Galeazzi (Karma County) on electric and double bass. I’m also in The Millionaire$ with them so we’re used to playing with each other. For some other tracks I get Hamish Stuart – different folks for different strokes, to coin a phrase. I believe in getting someone who will really suit the track and let them interpret how it should be played, with a little direction from me. Jim is in for the rockin’ and country cruisin’ stuff; Hamish for the swingin’ brushes and the funk. I lean towards time keeper drummers – my heroes like Jim Keltner, Ringo Starr, Ricky Fatar, Al Jackson – players who support the song and make it sound better without drawing too much attention to themselves, and Jim and Hamish are those guys.
I met Stu through my association with the socalled ‘University of Jackie Orszaczky’. A lot of punters wouldn’t be aware of Jack but to people inside the biz he was a rolled-gold master and had a huge impact on Australian music – as a peerless bass player, arranger, band leader, composer, entertainer. His decade-long Tuesday night residency, originally at The Rose Of Australia Hotel in Erskineville and later at the Newtown RSL, is the stuff of legend and was once cannily described as ‘the best free Tuesday night of entertainment anywhere in the world’. Jackie fostered a couple of generations of musicians in his various ensembles – great drummers, bass players, horns, everything. I tapped into this milieu of talent early on and have been using folk like Stu, Anthony Kable, Dave Symes, Felix Bloxsom et al on sessions for nearly 20 years. MIXING CUES I’m on a deadline now. Through the good offices of Abbie Cardwell I can get So Said Muhammad Ali mixed in America by Nick DiDia who has mixed Bruce Springsteen’s last three albums and worked with everyone from Rage Against The Machine to Powderfinger – he’s a heavy hitter. This is a fantastic break but I have to get the files to him in Atlanta, GA, by Tuesday Feb 3 when he has a window in his schedule. It’s 2pm Friday Jan 30 and I’ve just finished tracking the horns. I get it to the Fed Ex man in time for the 5pm curfew. Nick sends me back stem mixes via FTP internet delivery and, guess what, they sound fantastic!
I spend the first week of Feb mixing all the cues. I do a full stereo mix at 24-bit/48k, this is the film and television standard. I also do stereo stem mixes, sometimes up to 10: drums, percussion, bass, guitars, pads, strings, keys, FX, vocals etc. The full stereo mixes are for the key film workers – the director, editor, producer, me, my publisher etc – so everyone has a carbon copy of what’s been recorded, we’re all in the loop. And it’s also for
downstream if there is any third-party usage like a soundtrack album. The stems are for the final film mix where the dialogue, foley and the cues all get mixed into the final soundtrack – what you hear in the cinema. The stems allow the director and the film mixer some flexibility in the final mix. If the track is too busy and some of the instruments upset the emotional content of the film they can be attenuated or muted. If any instruments physically clash with elements within the film – say, a cello is obscuring some female dialogue – again, it can be ducked or muted. As a composer I am conscious during tracking not to rub up against any audio components in the film but, hey, sometimes the odd one slips through and with stems you can fix it. Also the film mixer can change any panning information and add effects, which he does in this case – I tend to mix a little dry so a bit of reverb gets slapped on. Mix-wise I use a console and outboard gear – no plug-ins. I had a revelation a couple of years ago. I dusted off an old Ibanez AD-230 analogue delay that drummer Felix Bloxsom left behind when he went to live in LA (thanks mate) and it really lifted my skirt. I’d been using all sorts of plug-ins and been lulled into the complacency of convenience but I’d forgotten how good a funky old delay unit can sound – it was wide and warm and completely musical. I’ve never liked mixing in the box. I like to spread a mix out on a desk, give it some real estate, get my hands on the knobs, walk around the room eating a sandwich instead of looking at a screen. All the fader automation I did in ’Tools but the rest is old school – outboard compression, TC reverbs (mostly plate patches) and delays – but no bus compression and no limiting, I leave that to the grownups at Soundfirm who do the final film mix. FINA L STRAIGHT In February I attend seven days of the film mix, mostly to be on hand if a mix is in the wrong place or if any files are missing, which can happen – better to be looking at it than looking for it. On some cues a few stems get dropped out of the mix for various reasons, in some instances cues get dropped completely. This is when you have to dissociate yourself from the music, you can’t afford to be too precious – your music is there to serve the film. I put my hand up once about an omission, mainly because I felt it was a vital link to a cue further downstream, part of a trilogy that establishes the lead male character, but ultimately to no avail.
In August I go to the cast and crew screening in Sydney. Since the last time I saw the film it has been re-edited, possibly more than once. Strangely, some music is back in, some other bits are missing and some cues get repeated in other scenes. That’s show business. The mix sounds good in the theatre and my name is spelt correctly in the credits. When’s the next one happening? Call me, I’m your music guy, okay?
AudioTechnology THE MAGAZINE FOR SOUND ENGINEERS & RECORDING MUSICIANS
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MUTE OR MOOT?
A subjective Analogue vs Digital comparison. Just when you thought it was safe to go back in the water…
Text: Chris Vallejo
Please bear with me – don’t turn the page assuming this will be yet another pointless technical discourse or wordy opinionated blather on the anachronistic analogue-versus-digital debate. In all honesty, I’d like to think that argument is well and truly dead by now. Instead, I’d like to propose a purely subjective, interactive and hopefully enlightening experiment: a practical comparison between two legitimate methods of audio recording and mixing with no real expectation of a winner or loser. Before we dive in, however, I must make a confession: I used to be a diehard analogue fan that could see no other way. I resisted buying into digital for as long as humanly possible – preferring somewhat foolishly to lose business – recording the old fashioned way to ferrous-coated plastic rather than to ones and zeros. To me, digital recording was the dark side; the cheap and nasty; the reason modern records sounded terrible; the unreliable amateur prosumer format. In short, I had a thorough, unadulterated hatred of modern recording technology. I’m not entirely sure where this hatred stemmed from, although I do recall using an ADAT once to make a record and the result, to my ears, was torturous: thin and sharp, like listening to your mother-in-law screaming through a megaphone. Of course this had nothing to do with the inexperience of the operator, or the cheap preamps, or the untreated room, or the handful of AT 10
quite ordinary microphones! Of course it was digital technology’s fault! HUMBLE PIE Move forward 10 years and I now eat humble pie every single day. I would now say that approximately 85% of the projects I’m involved with use digital technology in some form or other. Typically, the ‘talent’ is tracked to two-inch tape (preferably 16 track) and once the beds are done, the whole lot is transferred to the realm of ones and zeros via Apogee AD16X converters. Once ‘In the Box’ (ITB), edits and comps are executed where necessary and the last remaining overdubs added before the songs are mixed through an analogue console with analogue outboard. I’m always mindful to ensure where possible that the DAW is spitting out audio at unity gain and I rarely use plug-ins, preferring outboard compression, EQ, reverbs and delays etc.
This is my preferred working method, particularly given that most bands aren’t prepared to purchase a box or two of two-inch audio tape at $450 per roll. Working this way I get the sonic benefits of analogue tape during tracking, with the benefits of digital storage and editing. Sure, it might not suit everyone but it seems to work well for me, and I consider myself humbly and mildly successful in the recording game because A: I’m still in business and B: it’s all I do…
TRACKING SETUP Talent Band: Traps Guitar: Kieren Day, Ben Hassell Bass: Matt Edge Vocals: Kieren Day Drums: Mal Page Analogue Multitrack: Ampex MM1200 two-inch 16-track. Console: 32 Channel/8 bus AWA P66500. Mixdown recorder: Ampex ATR102 ¼-inch two-track. Tape: Quantegy GP9. Digital Converters: Apogee AD16X and DA16X w/ Apogee Symphony System. DAW: Logic Pro Version 8.02 on a Macbook Pro. Bit depth/sample rate: 24-bit/96k.
I had a thorough, unadulterated hatred of modern recording technology. I’m not entirely sure where this hatred stemmed from…
As I use both methods interchangeably, I started listening out for the subtle nuances that separate the two, but would often find myself listening to one medium without a solid and reliable reference to the other. I soon began wondering if the differences were as significant as my preconceived ideas were telling me. And, of course, I’ve seen the myriad shootouts and comparisons here, there and everywhere but am forever suspicious of their motives, particularly when they’re sponsored by software companies [Chris Vallejo, conspiracy theorist – Ed.]. But what if I did my own ‘shootout’ comparison, one that wasn’t sponsored by a biased third party, where a winner wasn’t the aim of the game? Would there be one in that situation? ANALOGUE’S ANONYMOUS Inspiration for this experiment came from contemplating the old recording, mixing and mastering labels that were put on early CDs, which were either: AAD, ADD or DDD. For those of you who have no idea what these labels stood for, the first letter represented the recording capture method (Analogue or Digital); the second was the mixing method and the third the mastering method – which I guess implies that vinyl records were all analogue (or AAA). I would always listen to CDs with that in mind, but the biggest difficulty was that you could never compare the same band, same song with a different recording topology. And I will always remember the standard CD sleeve caveat: “The music on this Compact Disc was originally recorded on analogue equipment. We have attempted to preserve, as closely as possible, the sound of the original recording. Because of its high resolution, however, the Compact Disc can reveal limitations of the source tape.” ‘Or, the other way around’, I would often quip to myself.
So what I proposed to our AT editor recently was this: Take a band into the studio for a day – preferably a good band with a catchy tune – set them up, make them comfortable, fire up the mirror ball and record the song the traditional way (i.e., to two-inch tape). Once they’ve knocked out a few takes they’re happy with, using the same complement of mics and preamps, record another series of takes, this time
straight to digital. There would be no ‘splitting’ the signal after the mic preamps and sending it to both formats. Doing two separate versions of the same song would allow each channel of the recording to have its own modified gain structure best suited to each medium’s strengths. With the analogue version, only the technology available to tape machines would be used. In our DAW recording, the use of a grid, comp’ing and editing if necessary would be permitted. Sure, the takes won’t be exactly the same, but who cares? (I can hear the naysayers already... put your hands down please and wait!). The idea is that the two takes would be damn similar if they’re done in the same session 20 minutes apart. I didn’t want to split the channels post-preamp because that introduces other problems such as gain structure and impedance issues that I wanted to avoid. I always find I run to tape slightly hotter with more margin for error, digital overs meanwhile are a no-no and hence gain settings need to be more conservative. Plus, why not make the experiment a workflow comparison of both methods at the same time? Then once tracking is complete, take each version and mix it in the respective domain – ‘version analogue’ through a console and outboard gear to an Ampex ATR 102 analogue master recorder, and the digital version ‘in the box’ using only digital plug-ins and the digital summing bus. Match them as closely as possible for levels and effects – i.e., if the bass is compressed and EQ’d with outboard, do a similar thing a similar amount using onboard plug-ins. Please remember this experiment is not supposed to be thoroughly terrific scientific – I will attempt to lay out all variables, assumptions and methodologies so we get to subjectively hear two perspectives on one song – I no longer have a particular preferred method of working, and importantly, nobody is sponsoring me to write anything favourable about either method, so I just plan to write about what I hear. PREDICTIONS I thought it would be prudent to write down what my personal expectations of both methods would
Mic placements remained unchanged betweeen the two takes.
SIGNAL PATH Instrument
Sytek 1&2 bussed
AWA pre 19
Neve 1084 L
Rode Classic II
Universal Audio 610L
Neve 1084 R
Universal Audio 610 R
Focusrite ISA215 L
Focusrite ISA215 R
Quad Eight MM403 -> Urei 1178
Quad Eight MM403 -> Urei 1178
Siemens V72 -> AWA G58
Siemens V72 -> AWA G58
Guitar Ben 1
AWA pre 17 -> Distressor
Guitar Ben 2
AWA pre 18 -> Distressor
Guitar Kieren 1
Langevin DVC L -> Opto
Guitar Kieren 2
Langevin DVC R -> Opto
be prior to the session â€“ as the recording and mixing process might influence what I thought I would expect, so this is what I anticipate heading into the experiment. I expect: the digital result to sound surprisingly good; a clearer mix with less noise and distortion; the top end not to be as smooth and the bottom to be a little more defined and not so woolly. Overall, I think it will sound a little harder. I also expect the mix to sound more generic with less tonal separation between individual tracks. I expect: the analogue result to sound slightly more pleasing, with greater clarity, more harmonic content and more colour and texture; the top end to be a little nicer and the bottom to be bigger but less defined (i.e., softer); the individual instruments to sound more clearly separated (due to my preconceived ideas about mix buses) and not so generic. AROUND THE TRAPS Once the band was confirmed I set about planning the experiment in greater detail. Thankfully, I had recorded the band two weeks earlier in a paid gig, so their setup was fresh in my mind and I still had the tracking sheets from that session handy.
Both guitarists like to use two amps each, and we found that we got a great sound just having them next to each other and miking both. When doing this I like to use a dynamic mic on the brighter of the two AT 12
amps, and a large-diaphragm condenser on the other. The choice of mics also wasn’t anything particularly specific, but again, I typically find that using different mics adds a little more dimension and breadth to the spectrum (as opposed to using four SM57s, for example). The drummer has a pretty nice mongrel kit consisting of a Dandy kick drum – vintage Australian, which I always like – and two Yamaha toms. To my great relief he was very solid and consistent so I didn’t find any need to use any compression while tracking. Bass guitar was recorded using a Beyerdynamic M380 dynamic mic into a Siemens V72 and an AWA G58 compressor. The Beyer probably puts out a little too much level for the V72 to handle comfortably but it produces a slight distortion that can be really pleasing and useful for the bass tone. The band was set up in the same room just like their rehearsals, which is my preferred approach except for the use of headphones instead of a PA. There were baffles set up around the drum kit and the guitar amps were placed behind these baffles pointing away from the kit. The bass guitar amp was placed directly in front of the drum kit as far away as possible facing in the other direction. Level out of the amps wasn’t essential for the tracking as everyone had headphones via a Hearback system, so the bass amp wasn’t cranked by any stretch.
Before the band arrived I demagnetised, cleaned and aligned the heads of the tape machine using an MRL for 520Wb/m2 or +9 with an overbias of 3dB. The machine is only a touch down at 18kHz and has a low-end hump at around 120Hz running at 15IPS. The tape spinning on the spools was Quantegy GP9 and the tape was bulk-erased using a Weircliffe bulk eraser. As for the digital setup, I decided to track at 24-bit/96k as I thought a major difference between the two would be the extent of the top end – the tape machine (and analogue outboard/console) goes above and beyond 20kHz, so I thought it prudent that we let the digital version extend higher than 20kHz as well. UP & RUNNING After a fairly relaxed 1pm load in, we were ready to start tracking at 2:30pm. A tempo was worked out and the band did two takes before coming into the control room for a listen, with comments. A couple of minor tweaks later, we were tracking the next four takes in succession before the band stopped for another listen. Take 3 was deemed the keeper, and it was marked on the reel of GP9.
Rather than dither (sic) about, we thought it would be best to go straight back into the recording room and track the digital version in order to preserve the same mindset and feel as much as possible. Changing from analogue to digital was a simple case of swapping 16 patch leads. I also got them to play through a chorus
of the song to work out what levels needed adjusting – I got overs on Tracks 1, 3, 7 and 8 so these were all lightened by about 5dB (or the minimum resolution on the preamps). We started tracking to a click and did three takes in succession. The band was content with Take 3 except for the ending, so this was comp’d from Take 2. The end solo was also revisited, but after a listen and vote on the alternate it was decided that the live solo was best, so it was kept. CUT & RUN Exactly 21/2 hours after we started, the band tracking was complete. We packed up drums, amps and mics before moving on to vocals. Kieren is a very competent and consistent vocalist and doesn’t require too many takes to get what he’s after. I also mentioned to him that the analogue vocal needed to be done on one track only, so anything untoward on the ‘best’ take would need punching in – not the Ampex MM1200’s forte by any stretch. Thankfully, all it took was two takes on analogue and three takes on digital and the vocal was complete. We left the studio at 6pm, exactly five hours after we arrived and headed to The Vic for a few self-congratulatory amber ales! SPOILER ALERT If you haven’t already downloaded the song from the AT website, I would humbly urge you do so now and listen to it before reading any further into the article. Maybe for larks you could even write down what you AT 13
I did get the sense that the placement of the tracks seemed to ‘sit’ more easily through real faders – but on the flipside, the digital mix felt much more precise.
expect the main differences to be. It’s important to do this because I’m about to describe what I think the differences are and what sticks out to my ears, and this may influence how you listen to the track and what stands out to you as being different. Listen to it a few times and concentrate on the subtle differences if you can, and try and determine which version you think sounds better versus which version makes you feel better. ANALOGUE MIX SETUP Instrument
API 2500 Bus
API 2500 Bus
AWA EQ -> 160VU
API 2500 Bus
API 2500 Bus
AWA EQ -> 160VU
API 2500/EMT 140
API 2500 Bus
API 2500 Bus
API 2500 Bus
UREI 1178 -> SPL Transient Designer
UREI 1178 -> SPL Transient Designer
Neve 2254 -> EQP1A
AWA G58 -> EQP1A – Dist CultVult/EMT/Lex PT
Guitar Ben 1
AWA EQ -Distressor
Guitar Ben 2
AWA EQ -Distressor
Guitar Kieren 1
AWA EQ -EMT 256
Guitar Kieren 2
AWA EQ -EMT 256
Two bus – Thermionic Phoenix Master Compressor -> Manley Massive Passive
I would also recommend, if you can cope with the bandwidth, that you download the uncompressed non-mp3 version (for Mr Massenberg’s sake if nothing else…). There’s a 24-bit/96k version on the AT site for those who really care, but 16-bit/44.1k should suffice. There’s a digital mix version, an analogue mix version (converted to digital obviously) and a concatenated version that swaps between each of the two every 15 bars (separated by a 1kHz pip). MIX TIME Okay, confession time. I originally did a mix soon after the session but when push came to shove writing the article, I had another listen and could already hear the chorus of protestations from both camps, Analogue and Digital. On listening – six weeks after the tracking session – I had decided that the mixes were adequate but a little too soggy and undercooked for such a potentially inflammatory article. I also thought that a more in-depth discussion was required about the mixes, and to be honest I couldn’t truly remember the process or issues I had faced when undertaking them. ANALOGUE MIX SETUP As per usual I started by normalling the desk, checking the alignment of the tape machine and making sure everything was labelled. I had also decided to set a limit of three solid hours per mix medium, as I felt the ease of everlasting tweaking on the digital mix ITB might give it an unfair advantage.
I started with the analogue mix, simply because it was a natural starting point, being the medium I was more comfortable with. If I mixed this first, then used it as an ITB reference for the digital mix it would more than likely give me the most promising result. Thankfully, the arrangement of the song was super simple; two guitars, drums, bass and vocal couldn’t really be any more bread-and-butter. I had also decided that I wouldn’t do any fader moves or automation as AT 14
it would just complicate matters, particularly given the self-imposed three-hour limit (plus I really don’t think the track needed it). A couple of points to note: There was a noticeable ‘print through’ of the vocal on tape, particularly because the vocal starts the song along with the bass. To eliminate this I used a Kepex gate set with a fast attack and slow release. Also, as there was no fader automation, I decided to keep the digital files unadulterated rather than top and tail them for noise where necessary. There was a mix of distortion added on the vocal and bass, both done in parallel with a Thermionic Culture Vulture. I knew that this effect would be repeatable within Logic using auxiliary sends, and I tried to avoid effects that didn’t have a digital equivalent (the Ursa Major Space Station being one of them – using that thing on guitars is just cheating)! The whole mix was bussed through a Thermionic Phoenix Mastering Compressor and a Manley Massive Passive with a few tweaks to the bottom and top end, as well as a slight dip at around 350Hz. The console’s limited low-end EQ (nothing between 250Hz and 1kHz) often results in a build up of frequencies in that region so I often cut a little on the overall mix. After three hours of tweaking, poking and prodding, I thought I had come up with something that felt relatively good. There was also a rough mix and a previous mix to compare with, and all in all I was pleased with where this mix was placed. I will have to confess that I like mixing in the analogue domain – it’s just so intuitive and easy. Moving the faders, patching the effects and tweaking knobs has a very satisfying and tactile sensation to it, and I like being able to do two or three things simultaneously – not having a control surface on my Logic setup means everything must happen with a mouse push or button press – one tweak at a time… slow. THOUGHTS & IMPRESSIONS To my ears, good analogue equipment has a very wide ‘sweet spot’ – you can move faders and knobs around and in many cases it’s a matter of finding the point at which everything sounds best amongst settings and points that already sound good. Admittedly, it’s hard not to go for the stock-standard signal paths that past experience has favoured: AWA Vari-MUs on vocals, ELI Distressors on guitars, dbx 160VUs on kick and snares etc.
Once the analogue mix was printed to tape and transferred back to the digital realm, I set about matching the levels of that mix ITB. I imported the analogue mix so it was easy to reference to what I was working on. First thing I noted was how clear the digital stuff was – no haze or halo around the sounds, just a more direct sound. As unscientific as that might be, the digital stuff just seemed to have an unadulterated clarity that was missing off tape, not to mention the absence of mild distortion and tape hiss that was present on every analogue track. I also thought this hiss and distortion might constitute most of the ‘warmth’ everyone bangs on about. I attempted to match as closely as possible the analogue mix settings, at least as a starting point for the mix, figuring that I would then tweak further depending on how it all fitted together. Something that immediately jumped out at me was how that ‘clarity’ I was enjoying so much when I first brought the mix up was making everything feel more disparate – like I was trying to mix oil and water together – whereas the analogue stuff felt like it fitted together more cohesively and effortlessly, like mixing ice cream into a milkshake. The next thing that stuck in my mind on mixing ITB was the narrower ‘sweet spot’ that the elements of the mix appeared to have – getting the vocal to sit right was just a little trickier and it felt like there was a very fine line between it sticking out like dogs’ balls, and being too quiet. I remember thinking on the analogue mix that I could move the fader a good centimetre or two either side of ‘good’ and it still sounded acceptable. Maybe this has something to do with the resolution of the physical faders vs digital faders, or maybe it’s the extra depth of analogue mixes… While I know we’re entering subjective territory, I did get the sense that the placement of the tracks seemed to ‘sit’ more easily through real faders – but on the flipside, the digital mix felt much more precise. On the analogue console everything feels roughly ‘around-about’ near where it should, but you’re never sure until you run some tone out again and check it with a goniometer. The other thing I noticed (and I only noticed when comparing the two) was that using the analogue pan controls gave an around-about movement of the image in the stereo field – but it was less than perfect. It’s more noticeable on an AWA console because you have to press a Pan button if you want to use a stereo pot, otherwise it’s only Left, Right or Centre – engage the pot, hear the scratchiness and
the image goes slightly broader and less focussed! Ahh, vintage electronics. As a former analogue diehard I will have to confess to being surprised at how close the two mixes were. In fact, I was hard pressed initially to prefer one over the other, as they both had their qualities. The digital mix had clarity and definition, although I didn’t think the drums sounded nearly as good as they did off tape. The analogue mix sounded more inviting and a little ‘bigger’ but suffered to my ears from a bit of smearing, where everything sounded a little more mushy in the bottom end (no doubt thanks to phase-smearing inductors, capacitors and transformers). One more thing that also stuck in my mind was how I couldn’t get the ‘power’ out of the snare or kick – in the digital mix they just sounded a bit papery and flappy without sounding like they had any sort of punch and thump. It’s almost like I could hear the voltage swing in the analogue version whereas digitally the signal was almost too accurately reproducing the air pressure flapping off the drum heads for it to sound exciting. To my ears vocals and guitars are probably clearer in the digital mix – the guitar amp sound is reproduced accurately with minimal artefacts and the amps manage to maintain their power and front, which really push through in the mix. Cymbals, however, don’t fare as well – the analogue mix cymbals have a sparkle and depth that the digital mix misses out on completely. Listen to the digital-to-analogue moment at pip 2:17 in particular – in my opinion the watershed moment. The difference when Mal starts smashing his cymbals in the solo is night and day. The same applies for the guitars too. Despite the fact that they are hardpanned in both mixes, they feel wider and bigger in the analogue domain. I would have to say that those artefacts are almost certainly the tape machine response as they didn’t sound that good when they were going down (tape machines are definitely in the business of flattery!). THE LONG & THE SHORT OF IT In many ways this experiment merely confirmed many of the suspicions I already had – that digital had come a very long way since the time I had made up my mind to hate it, but it’s not there yet. Considering the digital mix was done with a stock-standard version of Logic Pro V8 and a Macbook Pro, the results were nothing short of amazing.
To sum up, surprise surprise, I prefer the analogue mix. In fact, on further and deeper listening, I think it smokes the digital mix in emotion and attachment. It just feels better, sits better, has far more dimension and depth, the drums sound bigger and more exciting – with the digital mix feeling flatter, harder and colder. I do, however, prefer the snap and front of the snare in the digital mix, and the guitars and bass sound more present/immediate too although this is not necessarily a good thing. One more thing I thought I’d mention before I wrap this up – I level matched the two versions just by ear, and in the concatenated file, switching between the two, the level change is relatively imperceptible. However, when scanning each file for headroom, the analogue file has 4.45dB before clipping whereas the digital file has a mere 1.83dB – but to my ears the analogue version sounds louder. Hurray for tape machine response and soft limiting. If you have the means, something that might be enlightening is to let a non-techy friend or loved one listen to the mixes – explain briefly the concept but then leave the rest to them. You could well be amazed by what they can hear, probably because their perspective is less clouded by technicalities than those who read this magazine. In fact, people who don’t know the details of the experiment (and the whole analogue/digital debate) will probably go more on gut and emotion rather than the differences that you or I might perceive. This experiment has left no doubt in my mind that although analogue is the tried, tested and proven method, digital technology is here front-and-centre and the gap between the two mediums is narrower than ever. It just makes me wonder if we’ll ever be doing this comparison in the future, and if we did, would we have a clear-cut winner?
Ben Hassell Mal Page
Nevertheless, I’m not planning to sell my console, outboard and tape machines just yet. I find it eminently easier to mix using outboard and faders, and feel that the colouration you can achieve from all these different outboard topologies can’t be matched by a computer, no matter how advanced the modelling. The differences may be subtle, but maybe they’re too subtle for our conscious minds to really pinpoint – music appreciation involves a subconscious emotional response that is impossible to quantify with a mere black and white comparison.
You can download the mixes here: audiotechnology.com.au There is an analogue version, a digital version, and a composite version that swaps between analogue and digital every 15 bars (separated by a 1kHz pip), starting with the analogue. Do you think you can do a better digital mix ITB? Download the individual files from the AT site – and be sure to let us know what you think!
audiotechnology.com.au AT 15
HOME GROWN While not quite an endangered species, Jim Moynihan is certainly oneof-a-kind. AT flies east to investigate the layering of this colourful composer’s plumage. Text: Brad Watts
Earlier this year I was invited to pop into ABC radio in Melbourne for a bit of a chat on-air about the state of desktop recording in the 21st century. Of course I wasn’t the only member of the panel. My partner in this broadcasting crime was the inimitable Jim Moynihan, also invited on the show to offer a performer’s viewpoint on the world of computer-based production and performance. Jim had on hand a copy of his latest album, Zoomorphic, released via his own label, Omelette Records, and under the moniker of ‘Spoonbill’. So what’s with the ornithological theme you might ask? Well, it seems to be the common thread throughout Jim’s work, yet not even Jim himself seems to have a solid reason for his infatuation with birdlife. Suffice it to say his production skills have developed markedly since his fledgling years (oops) as a drummer and percussionist. Nowadays Jim, a.k.a. Spoonbill, regularly flies the coop (sincerest apologies) to perform at various international festivals. More recently I had the pleasure of spending a stormy Melbourne afternoon checking out the Spoonbill studio, discussing production techniques, and discovering some of Jim’s quirkier sound sources. Jim’s music defies description, but in the current music climate it could well fall under the banner of swamptronica, cabarabstract, or quirk-hop. It’s an interesting listen, and clearly Jim spends countless hours hatching (there I go again) his precision audioscapes. As is customary with any Home Grown interview in AT, we kicked things off with the usual scrutiny of Jim’s studio and preferred hardware and software combinations. After listening to the Zoomorphic album and hearing plenty of superbly crafted synthesis and sonic shenanigans, I was surprised to see Jim’s kit consist of remarkably little hardware. I assumed I’d be seeing at least a few pieces of outboard compression,
but no, Jim executes all his composition work within the confines of a Mac – ‘in-the-box’ as the working method is so well known. Brad Watts: So Jim, can you show me the tools behind the Spoonbill sound? Jim Moynihan: Sure… well, as you can see, it’s relatively simple. Everything is done in Cubase 4. I’d like to move to v5 to get access to the ‘vari-audio’ functions, but that means I’ve got upgrade to OSX 10.5, which I’ve been reluctant to do. The system works fine as it is on 10.4, and being mid-project I’d be mad to change anything. Audio gets to and fro’ the Mac with the Metric Halo ULN-2. The preamps on that are really good. When I’m looking for a really clean signal I just go straight through that and it sounds great. They’re excellent pre’s, way better for what I’m doing than the TL Audio unit on the desk over there [Jim pointing to a forlorn and dusty Ivory 2 Series preamp]. I’ve got a few friends that own some nice gear and we share stuff around, which is a great way to do it because none of us are actually recording the whole time. Plus we buy and sell stuff off each other. BW: Always a good tip: sell your gear to someone you know. JM: Exactly! SAMPLING STALWART BW: What else have we got here, and what on earth is that thing over there in the corner?
JM: That’s a Yamaha RS7000 – a sampling sequencer. It was an excellent bit of kit when it came out, and it’s still quite a good sampler today – I use it all the time. I bought it around 2002; I think that’s when it first came out. It’s 24-bit, which was unusual at the time. It’s been totally thrashed. I’ve taken it to I don’t know how
many gigs, literally hundreds and hundreds of them. It’s withstood enormous clouds of dust at festivals and it’s still functioning perfectly. And it’s got quite a good sound too. I guess you could say it’s the big brother to the RM1X sequencer, only with 64MB of sampling RAM. BW: And plenty of outs? JM: Yeah, I’ve got the expansion board in it so there are plenty of outs. I use it for performances still, and a little bit for production, but mostly for some of the sounds in it that I still find interesting. It has this thing called MIDI Delay, which I hadn’t seen on many things before. It’s a fun thing to play with, but now that I’ve got the Access Virus I’m using the Yamaha less and less. The Virus T12 is just awesome – that’s the machine which has been the most fun lately. BW: And what sort of MacBook are you running? JM: A MacBook Pro and a ton of Firewire harddrives. BW: Looking at that hard-drive stack, I’m sure you must have had your fair share of hard-drive hell… JM: Well it’s hard when you’re running a project like this with loads of tracks. I always copy the project over to the internal drive which works better for me. I basically use the external drives for sound libraries and backups. Otherwise it’s just too much for the Firewire bus to handle, with the Metric Halo on there as well. I should really add a Firewire card to the ExpressCard slot, but as it is things are running okay, even if the internal drive is only 5400rpm. The only drive I’ve really had fail on me was the Lacie Porsche drive, designed by F.A. Porsche. Unfortunately the thing fell onto a concrete surface and smashed to smithereens – no fault of its own. But the mad thing is that since the accident it’s come in really handy as a shaker; you can hear the platters shaking around in there. It’s a great addition to the studio that one! [See the AT website for a video demo of the hard drive shaker!] BW: That’s nuts! And the Focal monitors? I’ve yet to hear them to be honest. JM: Yes they’re a recent addition and so far I’m completely stoked with them. I auditioned the Adams at the same time and couldn’t go past the Focals – the Twin6 BEs with beryllium inverse dome tweeters –
Jim sporting the Spoonbill regalia.
they’re apparently good up to 40kHz. I think they’re probably a bit big for this room but I’m loving every minute of them. SPOONBILL TRACKS BW: Can you show me how you pull the Spoonbill tracks together.
JM: What I’ll usually do is start in a loop format; usually drums, then bass, then melodic single-hits. Every hit is a placed bit of audio – I don’t use a sample player plug-in as such. Then I work melodies in over the top using synth plug-ins or one-hit samples, constructing melodies out of a whole bunch of different sounds. I build my basslines the same way. BW: So you’ll have one note from this sound, another note in the bassline from another sound and so on? JM: Yeah. Rather than having a conventional bassline or melody, often the grooves are built from a whole bunch of disparate samples that come together to create the mood. That’s a fundamental aesthetic of the music. It’s a real collage cut-’n-paste sort of style of audio production. I can take it in almost any direction I want to because I’m using so many different sounds. Sometimes it’s the sound of a foghorn as a note, pitched in to fit, sometimes it’s a pitched-up kick drum. BW: So when you’re compiling this collage, and say you’ve got a bassline that’s already made up of six sounds and you’re looking for the last note, how do you go about auditioning the options for that final character? JM: If there’s just one note missing from a part I’ll probably audition sounds using the Virus, and I’ll usually bring in hundreds of other samples, import audio then audition heaps of sounds and choose the ones I really like.
WHAT’S OPERA DOC? It turns out Jim’s grandfather was one of Australia’s first industrial designers. Charles Furey founded Charles Furey and Associates in the late ’50s, before industrial design really even existed as a profession in Australia. Mr Furey was a pioneer of the industrial design movement in Australia and no doubt this influenced Jim’s decision to study design at university. There are various examples of Jim’s forays into design placed around
the Spoonbill studio. Mr Furey passed away on the 30th of May this year, aged 92, but his designs will inevitably endure, not least of which will be the stackable polypropylene chair. It transpires that Charles Furey was also responsible for the design of the original recording console installed in the Sydney Opera House, a picture of which hangs proudly in the Spoonbill headquarters, and rightly so.
BW: Have you got a fast way of doing that? Say, for example, you’re looping eight bars of bassline and there’s a particular note requiring a new sound, how do you keep track of it all? JM: Well, I’ll either audition a new sample live while the track is playing, or I’ll import a huge collection of sounds into the project, and just try putting things on different notes. Slowly I start building up a groove, then I’ll come back to it and decide if I want certain sounds in places and if I’ve got something on hand
One tiny controller covers all input basesfor Jim. AT 17
We’ve never heard a rattlesnake, but they sound like these things apparently. See the AT site for a demo.
Access Virus T12: the Spoonbill weapon of first resort.
The somewhat underrated Yamaha RS7000 sampler.
Cubase 4 is Jim’s vehicle of choice for alignment of his vast array of quirky sounds. Notice each and every percussion hit is placed lovingly into position as audio – not a sampler plug-in in sight.
Rather than having a conventional bassline or melody, often the grooves are built from a whole bunch of disparate samples that come together to create the mood.
that’ll fit. I might have to pitch-shift a sound or manipulate it somehow. Usually I build up a track starting with a drum kit or a set of sounds, perhaps three or four different sounding kicks. A lot of it is based on knowledge of what sounds I’ve got in my inventory, and what sounds I know will translate well in a performance situation and on a big sound system. BW: So you’ve definitely got your favourites then? JM: Absolutely. And that’s developed over the years, just working with my libraries and hearing what sounds good. I think I usually start with drums because I’m a drummer. It evolves quite slowly to be honest. I usually spend a day or two just on the main loop – getting it nailed and building the density up until it’s full and deep sounding before I even start on another section. Once I’ve got a couple of sections ready I’ll go back and break it all down again and sketch out a composition. Then of course I’ll also record my own sounds. I love collecting unusual and interesting sounding instruments, like this little kalimba. I love recording foley, things like crushing egg shells to add to a snare sound. These are great as well [Jim shows me two small magnetic ‘pods’]. On the last album I had about five tracks with these in them.
BW: What are they? And how on earth do you mic them? JM: They’re called Rattlesnakes. They’re a kid’s toy that I bought in London years ago, for one pound believe it or not – you can’t get anything for a pound in London usually. They’re strongly magnetic and join together when you throw them into the air – incredibly hyper-transient and bright. I recorded them with some beautiful Schoeps mics in France in a very quiet studio. [See Jim’s demo of the Rattlesnakes on the AT websssssite.] BW: It sounds like the entire process can be quite arduous. How long do you spend here each day? JM: Probably 10-12 hours I’d say, on average. When I’m working on an album, what I usually do is I’ll have a collection of tracks that I’m thinking are cool, but which still have got a long way to go before they’re finished. I’ll book a mastering session three months ahead of time and work towards that date. Then I’ll spend 14, 16, 18 hours a day in here just getting it nailed. BW: That’s a great call: book your mastering session and then you have a deadline. JM: Yeah, I don’t like letting myself down. If I commit to something then I’m going to do it, and I’ll work really hard to make it happen.
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ON THE BENCH
DIY… or what to do on the weekend. Text: Rob Squire
With the footy season finally drawing to a close, for some of us there’s now time available to turn our attention to other activities. For me, that means building some DIY audio kits over lazy summer weekends. It may come as a surprise to many that a person such as myself, who spends all week fixing pro audio equipment, would be picking up the soldering iron on a Saturday afternoon and putting together an electronics kit project. (It certainly comes as a surprise to the missus who has visions of projects involving spades, forks and dirt!). I’ve been building kit projects since I was a kid and the process of doing this has not only added to my understanding of electronics but has furnished me with numerous pieces of equipment that enjoy regular use. For many of us who spend – or would like to spend – the bulk of their time just using audio equipment, there’s still an attraction to building DIY gear, which goes something like this: ‘surely I could build this stuff cheaper myself!’ Unfortunately, it’s no longer quite that simple. More recently, the manufacturing and retail costs of semi-pro and pro audio equipment have plummeted, and in 2009 this price rationale is hard to maintain. Given this situation, what good reasons remain for pursuing DIY electronics? JOINING THINGS TOGETHER Well, for starters, it’s pretty reasonable to suggest that basic soldering should be a part of any audio engineer’s skill set. Being able to repair a mic lead mid session might just be the difference between capturing a performance while it’s hot and everyone downing instruments while the bass player drives across town to get a lead repaired or replaced. (In my experience it’s always the bass player who picks up these jobs). Whether you’re repairing or making up mic leads or other cables from scratch, this can be a good way of dipping your toes into DIY and at the same time improving your soldering chops. LEARNING HOW IT WORKS Putting together a DIY electronics kit can also be an excellent way of deepening your understanding of how things work. This is particularly so in the case of projects issued by electronics magazines where, not only are the assembly details presented, AT 20
there’s also a detailed description of how the circuit works. For many people who end up making careers out of electronic design or repair, such circuit descriptions have often formed the starting point of their education. SATISFACTION There is no doubt that an extra degree of satisfaction is enjoyed when you’ve pulled a killer guitar sound using a mic pre you’ve built yourself. At this point it is also probably worth dispelling one significant and potent myth: that a piece of DIY gear can’t be as good as something you buy off the shelf. Today there are all sorts of kits available for DIY construction that will rival the quality of anything coming from a manufacturer, and this point shouldn’t be understated. If you have the time, reasonable soldering ability and can follow instructions, you can build a world-class piece of audio gear. Alongside of this, there are also some items of equipment you can build from a kit that simply aren’t readily available or affordable off the shelf. EXAMPLES & POSSIBILITIES Rattling around in the bottom of my tool bag is a small audio frequency oscillator. About the size of a guitar stomp box and battery powered, it’s one of the handiest additions to my tool kit. Producing sine waves from 40Hz to 18kHz with an adjustable output level, it saves loading my 20kg test set into the boot of my car every time I head out on jobs where I don’t need the precision and extra features my ‘professional’ audio oscillator provides. The kit for this unit costs less than $30, and depending on your level of expertise, could be put together in an afternoon. One fortuitous aspect of the design of this particular unit is that, with the level control up full, the unit generates a signal level of 4.5dBu, which in many practical situations is accurate enough to be considered your +4dBu reference level. Being battery operated the unit is ‘earth free’ and so while its output is unbalanced it will happily drive both balanced and unbalanced inputs without creating hum or earth loop issues. Naturally, I can’t help myself when it comes to building kits, and always find a way to modify the circuit to better suit my specific requirements. In the case of the oscillator
being so tantalisingly close to +4dBu output level, I’ve made a couple of small changes to the unit.
There is no doubt that an extra degree of satisfaction is enjoyed when you’ve pulled a killer guitar sound using a mic pre you’ve built yourself.
By simply adding two resistors to the output level control, one on each end of the volume control, I now have a level control that ranges between –40dBu and +4dBu, rather than the original range. This means if I want a typical mic level to check out a preamp, I rotate the level control fully counter clockwise and I know I’ve got –40dBu. If I want to check out a line level signal path I rotate the level control hard clockwise for +4dBu. I also installed a male XLR socket onto the case rather than the supplied RCA – generally speaking, a much friendlier connector in the pro audio world. This is a perfect example of one of the great things about DIY. As your confidence and abilities increase you can start to customise your project to produce a unique unit that exactly meets your requirements. WHERE DO I GET IT? Historically, electronics magazines were the fundamental resource for development of DIY projects with the actual kits being released through either electronics retailers or a few small enterprises that were set up specifically to supply the kit builder. In today’s PEA (post internet era), however, the options have broadened considerably with a number of companies producing high quality kits aimed fair and square at the recording studio operator and musician. In the world of kits there are essentially two types supplied: one called ‘short form’ and the other, ‘complete’ or ‘full’. A short form kit contains just the printed circuit board (PCB) and components to solder to it. With these kits you will need to supply a case, machine any holes required and sometimes add additional extra parts (or kits), such as power supplies, to produce a finished unit. Complete or full kits, on the other hand, supply everything you need to produce a finished unit, and these often include highquality pre-machined cases with whatever labels are required printed or engraved on them. Many of these full kits yield a finished product that not only performs as well as any product you can purchase but look as good to boot.
Without trying to pull together an exhaustive list, some of the suppliers of these kit projects are: JLM Audio: an Australian business owned by Joe Malone, that supplies high quality kits mostly in short form but also some complete kits. JLM specialises in mic preamps, DIs and power supplies. www.jlmaudio.com
This is where the fun begins. Turning this daunting pile of parts into a working piece of audio gear is a real challenge.
Elliott Sound Products: another Australian providing designs and raw PCBs only for a broad range of audio and non-audio projects. sound.westhost.com
Silicon Chip: an Australian electronics magazine that designs and describes all sorts of electronics projects, including many audio projects. While it doesn’t supply kits directly, many of its projects are available through Altronics and Jaycar. To give you an idea of the sorts of projects developed by Silicon Chip and available from Altronics, there’s the aforementioned audio oscillator, a very high quality headphone amp, stereo power amps and, just released, a stereo D/A converter that will beat the pants off many you can buy off the shelf. www.siliconchip.com.au
Hamptone: tube and Fet mic preamp kits. www.hamptone.com
PAiA: have been around for a long time producing kits especially for the experimental musician… need a Theremin, for example? www.paia.com
Fivefish: produces a number of kits specifically for 500-series lunchboxes. The 500-series lunchbox is rapidly becoming the goto format for DIY audio kit designers mainly because once the case is purchased, you’ve got a fully prepped standard format into
which projects can be slotted. All power and I/O connectors are ready to go and the lunchbox power supply can cater to more than one module. www.fivefishstudios.com
Seventh Circle: these guys pose the rhetorical question: “Who makes the best preamps for the money?” and then supply the answer: “You do!”. The company specialises in high-performance preamps based around some of the vintage American and British classics. www.seventhcircleaudio.com
BYOC: ‘Build Your Own Clone’ produces kits that replicate some classic guitar pedals. This is a good stop-off point for guitar players exploring DIY. www.buildyourownclone.com
WHAT DO I NEED? Fundamentally, the two main things you need to test the DIY waters is time and enthusiasm and it won’t work trading one off for the other! Beyond that you’ll need some tools with a basic kit consisting of a soldering iron, sidecutters, long-nose pliers, multimeter and a set of screwdrivers. Probably one of the hardest aspects of DIY is finishing the unit off by installing the circuit into a case along with all its required connectors and controls. As you’ll soon discover, machining a case yourself to house the DIY circuit involves more specialised tools such as a drill, drill bits, nibblers and the like. If you opt for a project that doesn’t come supplied with a pre-machined case you’ll quickly learn where all the time, effort and end costs go into manufacturing a product! STARTING SIMPLE It always make sense to start out on a project that’s within your abilities and since a good soldering technique will be essential to the success of any project the obvious staring point is to make up some cables before moving onto a small and simple PCB-based kit. A clear guide to soldering can be found at www.solderinguide. com. A magnifying glass and good light can be a bonus to enable close inspection of the solder joints to confirm that there’s a good flow of solder between the component and the PCB pad. Dry solder joints, where the solder is sitting on the component as a ball rather than flowing onto the PCB pad, will result in either incorrect operation of the circuit, or arguably worse, intermittent operation.
When assessing a kit project you must assess if the presentation of the kit supplies the amount of construction information that meets your abilities. Different kit suppliers will assume a certain level of electronics understanding and it’s important to ensure that your current abilities match the level of detail supplied with the kit. By all means push the envelope; after all, one of the good reasons for embarking on DIY is to gain knowledge and an understanding of what’s going on behind the knobs. However, it’s equally important to make sure the job’s achievable. SAFETY LAST Any project that involves installing mains transformers, wiring and ultimately connection to mains power should only be undertaken if you clearly understand how to do this safely. Most kit projects that use mains power very clearly explain how to wire up this part of the circuit and deviating from the explicit instructions, including any mechanical assembly, is a risky business. For these projects it’s always a good idea to get your work checked over by a qualified technician and tested and tagged – it’s the law after all.
From making your own mic cables to building a compressor, the options for DIY audio are wide open. And whether you’re ultimately looking for a new career path or the simple satisfaction of building something yourself, it can pay to look beyond what is available off-the-shelf to what you can create to put back on it! Which reminds me… I’ve got to go and put another coat of stain on the bookcases I’m building. AT 21
REVIEWS: IN BRIEF
BBE D82 SONIC MAXIMIZER During the 1990s, processors loosely referred to as ‘enhancers’ were all the rage. These devices were typically single-rackmount stereo units with a few mysteriously labelled controls for ‘low’, ‘high’, and often, ‘amount’. BBE was the big name at the time, with its Sonic Maximizer being one of the most popular devices in the enhancement genre, alongside Aphex’s Aural Exciter, which has been in production in various guises since 1975. This century, of course, has seen most hardware audio contraptions reincarnated as software and the BBE Sonic Maximizer hasn’t escaped assimilation. It’s now available as a plug-in from BBE itself.
The D82 plug-in as it’s called is compatible with all the usual native plug-in formats: AU, RTAS, and VST, so you’ll have no trouble installing it in either Windows or OSX operating systems. Amusingly, the D82’s cardboard package contains a single CD that, when opened, provides nothing more than a link to BBE’s website to download the latest version of the plug-in. It’s probably not a bad policy as the prospective user won’t be installing an outdated version, but it does seem a little crazy to have no software on the disc. Be that as it may, once up and running, operating the D82 plug-in is a matter of taste – wind in ‘Lo Contour’ for some extra bottom end, and add some ‘Process’ to hype up your high end. The third remaining control adjusts output level – it’s all very simple and completely self-explanatory. In use I’ve never found these devices to be flattering to stereo mixes, despite the claims of manufacturers. While the process may sound attractive at first, comparing against a well EQ’d version of your twotrack mix will reveal the hyped and ‘excited’ process to be just that – hype. Where these processors do come in handy is adding a little pizzazz to stereo keyboards and samples within a mix; even then it’s advisable not to get carried away with the effect too much – a little goes a long way. Sound effects for film and television can also benefit from a little ‘excitement’ – in fact, this is probably the area where this plug-in will find its niche. Enhancers and exciters can also be useful for sprucing up DJ sets for club playback, but the bottom line is: use sparingly. Brad Watts Price: $175 F. Payton & Son: (02) 9439 1822 or firstname.lastname@example.org
BEYERDYNAMIC MCE55.18 People have their favourite lavalier mics for a whole variety of different reasons. For some, the size is paramount – the smaller the better (and shaving half a millimetre from the profile can make all the difference). For others, they won’t so much choose their mics as choose the wireless system and use whatever comes with the beltpack. Water resistance and, of course, the sound quality and price are considerations as well. But what about a nifty squiggle mount? When I saw photos of Beyer’s (what’s officially known as) helical mount I could instantly see the possibilities. These mounts come in three sizes and will attach themselves to just about anything. What these squiggles do is effectively turn the lapel mic into a go-anywhere omni bud mic. No instrument is safe! Strings, flutes, horns… the MCE55 will happily attach itself relatively unobtrusively to all of them.
The mic itself sounds wonderful – open and natural, just like a high-quality omnidirectional microphone should. The frequency response is very flat with a gentle 4dB lift from 7kHz upwards. For wired applications the MCE55 has an optional phantom-powered preamp. Plug the mic’s mini fourpin XLR connector into the preamp (not much bigger than a standard XLR housing) and then it’s off into the desk, stage box or recorder. It’s the preamp that frees you from the restrictions of wireless beltpacks, including Beyer’s well-regarded Opus series. The mic itself is small enough to sneak into the hairline for theatre applications and will thrive in traditional lapel mic duties. But team the MCE55 with its merry band of helical friends you’re not so much getting another lavalier, you’re getting a genuine ‘small diaphragm’ omni that’ll happily slot into just about any situation. The word is, the squiggles will also be sold separately. Definitely worth stocking up on. Christopher Holder Price: $445 (includes helix mounts but not the preamp) Audio Telex: (02) 9647 1411 or email@example.com
EAR MONITORS AUSTRALIA A1 GENERIC IN-EARS There’s a real knack to wearing generic in-ear monitors. The key is to always wrap the cables up and over the top of your ears to ensure the earpieces themselves aren’t required to double as anchors – otherwise they’ll unusually fall out in short order. The weight of the cable must be borne by the top of your ear rather than the pressure of the earpiece in the ear canal, otherwise you’ll eventually find yourself on stage without foldback… and no-one wants that. Coincidentally, I’ve noticed all the contestants on Australian Idol are currently wearing the EMA A1s and hardly a single one of them seems to be fitting them correctly!
Ear Monitors Australia’s new A1 Generic in-ear monitors are a nicely balanced, well-constructed and tough new generic featuring a lightweight driver that offers good tonal balance. Supplied with a zip pouch, three sets of silicon eartips and a single set of foam tips, the EMA A1s will cater to most ear shapes and sizes. But like all generics, the key is to learn how to fit them in your ears and make certain the ear tips you choose are the right size to ensure a comfortable fit. My initial experience of the A1s began with me using tips that were too large for my outer ear canal. This forced me to push them in under too much pressure, which meant they simply popped out the moment I moved or opened my mouth to sing. Replacing them with the smallest of the three silicon sets has allowed me to insert the earpieces more easily, and they haven’t fallen out since. The temptation is to instinctively go for a bigger tip out of an expectation that it will be more secure, but this is a mistake in my experience. Like all good in-ear setups the tonal balance, or more specifically, the bottom-end response is critically dependent upon how well an in-ear monitor seals inside your ear canal. A poorly fitted in-ear will almost certainly lead to weak or non-existent bass response. The silicon eartips of the A1s seal well and provide reasonably good amount of bass, although I was still searching for a whisker more low end even after a tight seal was made. A bit of low end shelving EQ soon sorted this out, however, and the results have been impressive for a generic. Andy Stewart Price: from $270 Ear Monitors Australia: (03) 9844 2524 or www.earmonitoraustralia.com.au
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APPLE LOGIC PRO 9
Apple’s new Logic Pro has the ability to pull audio files into any shape imaginable. Text: Brad Watts
This has been a hotly anticipated upgrade, albeit not surrounded in the hype and suspense that accompanied the release of Logic Studio (v1) and the incumbent Logic Pro 8. At the heart of Logic Studio 2 lies the focus of this review: an upgraded and reinvigorated Logic Pro – now sporting the number 9 Guernsey. Logic Pro is, needless to say, the primary reason why people buy Logic Studio, and the primary focus of this review. The DAW and sequencing program, initially developed by Emagic in Germany, was acquired by Apple way back in 2002 to lead its charge into the professional world of music composition and editing. Logic Pro is now a fully-fledged member of Apple’s Pro Applications lineage, but the application hasn’t always had its plumage in such preened condition. Version 7 was Apple’s first attempt to assimilate the application into the grey-on-grey aesthetic of the company’s other white-collar applications, and was, in all reality, little more than a costume change. In terms of features, the application seemed much the same as v6. The next installment, Logic Pro 8, promised a lot, and did arrive with a truckload of revolutionary features, including a streamlined, single-window working environment that went a long way toward simplifying what has always been a complex application. With v8, Apple had time to shape Logic Pro’s internals to offer a vastly more consolidated workflow. This won the application plenty of new friends, as did discontinuing the XS-Key copy protection. But for many users of v7, Logic Pro 8 wasn’t the step forward it was promised to be. For some, the departure from established working methods accrued over seven iterations of the program was a little too much to bear, and there are many who still work within the so-called ‘confines’ of Logic Pro 7. To be frank, I wasn’t a big fan of v8 either, mainly because I was using it with Digidesign HD hardware, and the two historically harmonious bedfellows were starting to get a bit cranky with one another. More importantly, Logic, in all its guises, is a much happier camper when running in what’s termed ‘Universal Track Mode’. That is: using interleaved stereo files. These aspects rendered Logic Pro 8 a precarious platform for me, and I too eventually reverted to v7 for both stability and familiarity’s sake.
Apple have covered most amplification bases with Amp Designer, and there’s pedals aplenty within Pedalboard.
That said, there are plenty of Logic Pro 8 users out there that are extremely happy with its performance, and I’ve no doubt I’m one of the last to depart the sinking ship that is the Digidesign HD/ Logic Pro platform. Yes, that’s right, I’m ditching the Digidesign HD hardware.
As I mentioned in my last Mac Audio column, I’ve been flitting between ProTools 8 and Logic Pro 8 for some time now, never feeling all that settled. Things would have been easier for me if ProTools had been able to read interleaved stereo audio files without converting them to split stereo files, or if Logic Pro had been able to perform all its tricks with split stereo files. But I was daydreaming if I thought either of these wishes was likely to eventuate, and perhaps I should never have persevered with such a rig. But I’ve always loved the fact that latency was never an issue with the HD hardware. Incidentally, there’s now a millisecond latency value displayed in the audio preferences of v9, making it much easier to compensate for the inherent native latency issues. This point aside, the main problem with my Logic Pro/Digidesign HD time-trap was transferring projects between this system and the Logic systems of those I worked with – the process often became more trouble than it was worth. Plus, there were just so many functions in Logic Pro that wouldn’t function under the Digidesign HD command. So, as mentioned, the time came for me to bid Digidesign HD hardware adieu. THE PLAYING FIELD So, to keep you abreast of the system I’ve settled on, and to give you an idea of the systems on which I’ve been running Logic Pro 9, here are the current Apple computers I’m running. First up is my ‘work’ machine: a 2.4GHz MacBook with 4GB of RAM and a 500GB hard drive running OSX 10.5.8. The audio interface I’m using is either a MOTU 828mk2, a Digidesign M-Box Micro, or whichever Firewire or USB audio interface I’m reviewing at the time. This is a supported system, according to Apple, and it meets the minimum requirements as posted on the Logic Studio web pages at www.apple.com. Those prerequisites include an Intel processor, 1GB of RAM, OSX 10.5.7 or later, and QuickTime 7.6 or later – all easily attainable stipulations for an Apple laptop released a little over a year ago.
The other machine is my primary recording and mixing machine: an older, recently bequeathed dual 2GHz G5 running an RME HDSP 9652, a couple of ADAT eight-channel I/O units and a Universal Audio UAD-1 card. This machine was initially acquired to run my PCI Digidesign HD hardware but will now be running only the hardware listed above (I’ll probably track down a second UAD-1 card). Anyhow, the point is the G5 uses a PPC processor and is consequently becoming unsupported by Apple as shown by the advent of Snow Leopard. Logic Studio 2 did, however, install on the G5 quite happily and runs perfectly – the
software is a ‘universal binary’ and contains both PPC and Intel code. Obviously Apple isn’t in the business of making PPC-based computers any more so there’s no impetus to support the software on a G5, and certainly not a G4. The way forward from now on with Logic Pro is undeniably with an Intel-based machine, along with OSX 10.6 ‘Snow Leopard’ – which I’m about to install on my MacBook, having already seen Logic perform well on OSX 10.5.8. Be aware that it’s plausible you’ll not receive a great deal of support from Apple if you’re intent on running Logic Studio 2 on a PPC-based Mac. To ascertain the extent of Logic Studio 2’s backward ‘installability’ I attempted to install the software on a G4 iBook – it did install, but not before informing me that the video card held less than 64MB of RAM and that it wasn’t compatible with Apple’s OpenCL graphics-card-assisted processing. It works, but it’s unlikely you or the laptop will enjoy the experience. BOX & DICE Physically speaking, the Logic Studio package has become a shadow of its former self. The box is now nothing more than a CD/DVD style case with the nine install discs and a couple of brief manuals. The first of these covers the initial concepts and use of Mainstage, while the other, entitled ‘Exploring Logic Pro 9’ offers an introduction to Logic Pro’s primary features. The unexpurgated user manuals are available online only, either as online help files or PDF documents. Personally, I prefer to have a book on hand but there are obviously environmental (and marketing) considerations at play here too. The bottom line is that there’s now less paper, less shipping, and more online screen gazing required to bring you up to speed on the operational minutiae. YOURS FLEXIBLY Possibly the most publicised new feature in Logic Pro is Flex Time. Flex Time encompasses a number of tools specifically for warping audio files in both pitch and time, and is the evolution of features that have been hinted at since v8. To begin with there’s the ‘Flex Tool’, designed to drag and stretch audio files within a region to fit your required timing. There’s no need to slice or cut regions into tiny pieces before painstakingly moving and time-stretching them to fit. With the Flex Tool you can freely move audio in the time-domain, either elongating or compressing audio without affecting pitch. Set the Flex Audio mode to ‘speed’ and you’ll get much the same result as if you’d place the audio in a sampler and tuned it either up or down. This does work quite well but it’s easy to come unstuck so make sure you’re working with a copy of the original file.
As regions are analysed for their properties before applying any Flex Time adjustments, Logic Pro 9 finds and marks each transient within a region, and these are then editable within the Sample Editor. This allows another feature of Flex Time to become a reality: the quantising of audio. Audio quantisation can be applied over multiple drum tracks as long as the tracks are grouped and of the same region length. Once these criteria are met, the tracks can be phase-locked within the Group Settings window so one track can be referred to as the reference for phase alignment. Another superb new function is the ability to move notes or hits within an audio file using ‘Slicing’ mode. When clicking on a region with the Flex Tool, the Flex Audio dialogue box asks which style of Flex Audio you wish to use. Selecting Slicing mode analyses and marks transients, and facilitates the free movement of audio between two transient markers forward or backwards, without affecting any transients or the length of the audio being edited. What’s cool is that selecting two or more regions that have been analysed using Slice mode, lets edits made in one region simultaneously affect all the other selected regions. In the case of editing drum takes with multiple tracks, again, set the tracks to a group and make sure the ‘Phase-Locked’ and ‘Editing’ check boxes are ticked. Now editing one region will affect the equivalent regions of other tracks assigned to that group. Another feature made possible with Flex Time is ‘Speed Fades’. Hidden within the Inspector’s ‘Fade’ section is the option to alter a region-based fade to become a Speed Fade. This provides the questionably useful effect of a turntable slowing down or speeding up. Fine if that’s what you’re into, and I’m sure we’ll hear a spate of this over the next six months, but certainly it does sound like the real thing. Don’t go searching for this feature in any of the manuals, however, as it simply isn’t covered. VARISPEED Perhaps more astonishing and infinitely more useful are Logic Pro 9’s varispeed functions. Varispeed allows you to slow down or speed up an entire multitrack project according to tempo without affecting pitch, or affecting both speed and pitch as though the material was recorded on a multitrack tape recorder with variable speed. Adjusting the deviation from the project’s standard pitch and tempo is readable in percentage, resulting BPM, semitones or Hertz. What is truly stunning about this feature is you can slow down a multitrack project while leaving the pitch at its original tuning, record additional tracks (at the original pitch), then revert the project to its original tempo and pitch, and the newly recorded tracks will miraculously remain in tune with the original
I found it pretty easy to shift drums around and correct ‘drummer malfunctions’ without once chopping up a region into painfully tiny sections – the ability to also do this across multiple tracks is just brilliant.
Swing it high. Swing it low. This chariot travels at all speeds and any pitch.
Flex Time: You can now push and pull audio around to the project’s content, all within the Arrange Window. AT 25
recordings. Pretty clever, and super-handy for nailing that part you can’t quite play at speed. Interestingly, tempo information is now saved within audio files recorded in Logic Pro. This made an appearance in v8, but v9 adds the ability to import and export tempo information from any audio file recorded in Logic Pro, or an audio file having been subjected to Logic Pro’s tempo analysis features. Using Flex Time’s Slicing feature, v9 can ascertain the tempo of a region and will consequently speed up or slow down the region to suit the current project’s tempo – all the while keeping the original pitch intact. The effectiveness of these pitch and tempo transformations is very much subject to the original audio file quality. If you’re using poorly recorded, noisy or low-resolution files, or audio that is somehow laden with extraneous artefacts, you’re bound to end up confusing Logic Pro’s Flex Time algorithms and winding up with some very weird sounding and unusable audio. As always, verify the results by ear. Having said this, I’m impressed with many of the results I’ve had with the software in the short time I’ve been running it. The new algorithms aren’t up to the standard of the venerable Pitch ‘n Time plug-in from Serato, but they do a very good job, and are far superior to v8’s time-stretching smarts. Together, these functions are obviously Apple’s answer to Digidesign’s Elastic Audio technology, and go a long way toward bridging that gap in Logic Pro’s bag of tricks. I found it pretty easy to shift drums around and correct ‘drummer malfunctions’ without once chopping up a region into painfully tiny sections – the ability to also do this across multiple tracks is just brilliant. SOOPER PRODOOCER Logic Pro 9 incorporates other additional features that Apple likes to call ‘Production Tools’ (one wonders what the rest of Logic Pro is for). First of these is ‘Selective Track Import’, where entire tracks including regions and channel strip settings can be imported from other projects – drag tracks in from a project that was recorded at a different tempo, and the freshly imported audio regions will correct themselves to the current project’s tempo.
The ‘Drum Replacer/Doubling’ feature does exactly as it alludes. Say you have a kick drum you want to replace with a sample, or perhaps you merely wish to add a bit of Wellington boot to a malnourished kick. Selecting a track, then choosing ‘Drum Replacement/Doubling’ from the Arrange Window’s Track menu brings up a tidy little window that will allow you to set the kicks to be replaced using a threshold level control. The pre-listen button then lets you audition kick samples from your EXS24 sampler library. Once you hit ‘Okay’, Logic Pro 9 renders an instrument track alongside the original audio file with an EXS24 utilising the chosen kick sample, all played in time with the original kick recording. There have been methods to do this in previous versions of Logic, but this is by far quicker, and the different samples are much easier to audition. In a similar twist, regions can be converted to EXS24 sample instruments: select a region, choose ‘Convert Regions to New Sampler Track’, and Logic Pro will set up a sampler instrument with each hit of the region mapped to a note in the EXS24 sampler. This is exactly the same trick Steinberg’s Recycle once achieved so elegantly. Convert Regions to New Sampler Track can also render an entire region to an EXS24 sampler, letting you quickly map entire phrases/regions across the keyboard of an EXS24 sampler. One new tool in Logic Pro whose debut was long overdue is the option to bounce audio regions in place. The emergence of this tool, dear readership, will be met with cheers and celebration from every quarter. It’s absence has generated more complaints from Logic Pro users in recent times than any other – these users have been forced to watch other DAW applications like Cubase perform this trick for a while now. ‘Bounce Regions In Place’: it’s a riveting concept, and does everything you’d expect from such. You can leave, mute, or delete the source region, bypass effect plug-ins, ignore pan and volume automation, and either include or exclude any effect tails that may be part of the region’s playback. AT 26
And while this feature may seem insignificant compared with Logic Pro 9’s phantasmagorical feats of time-stretching strength, ‘Bounce Regions In Place’ is likely to produce the largest collective sigh of relief from the Logic Pro party. This alone will save yours truly large portions of my life. Why it didn’t appear in v8, or even v7, is completely beyond me. LET THERE BE ROCK Another aspect of Logic Pro 9 that leaves me feeling just a little befuddled, despite how good it is, is why Apple puts so much effort into including guitar oriented plug-ins in Logic Pro 9. ‘Amp Designer’ is the guitar amp simulator that was promised with ‘Guitar Amp Pro’, a plug-in that was essentially useless. Amp Designer on the other hand is the last amp simulator you could possibly need: it actually sounds remarkably good.
Apple has ticked all the right boxes here, with on-board emulations of any name amplifier you’d care to mention: various shades of Fender, two Marshall options, Vox AC30 and 15, Soldano, Mesa Boogie in all its history, and an ‘op-shop special’ that sounds pretty cool when you put a decent cabinet with it. Yes, that’s right, cabinets are interchangeable, as are mic choices and positions. Choose from dynamic, condenser and ribbon, then mix and match amp heads and cabinets until... umm… well, you know – it’s all about tone. The thing that’s odd about Amp Designer is that each amp head has the exact same controls: Gain, three-band EQ, reverb, sync’able tremolo or vibrato, and master output. Obviously a Fender amplifier doesn’t have the same controls as a Vox AC15, but I can only assume this made the coding process simpler. While we’re on the subject of plug-ins, I’ll point out a new Logic Pro preference allowing plug-ins to be viewed at various magnifications. As you’re no doubt aware, plug-ins can be quite small when presented on a large monitor, and these days Logic Pro is best viewed on a 16:9 widescreen monitor. To compensate for this, v9 allows Logic plug-ins to be views at larger sizes, starting at 100 percent and going through to 200 percent. Nice one. MORE ON THE FLOOR No self-respecting guitarist would feel secure without at least a couple of effect pedals between their amp and guitar – save Angus Young who simply plugs his SG straight into a Marshall. The pedal-board plug-in offers most pedals you’d ever require, and most are modelled, both sonically and graphically, on well-known and well-used pedals. There are 30 plug-ins in total to keep things varied – and that includes very useful ‘Splitter’ and ‘Mixer’ plugins. The Pedal-board plug-in even has its own control surface available through Apogee Electronics: GiO is a bespoke audio interface and foot controller for Logic Pro 9, which we’ll have a more in-depth look at next issue. At this juncture all I’ll say is that they sound alarmingly good – so much so that I wouldn’t be rushing out to upgrade your current amp and pedal simulation plug-ins without giving these a once over. LOGIC PREVAILS So after a look through the new features in Logic Pro 9, I should point out some of the under-the-bonnet benefits and pitfalls of upgrading. Firstly, Logic Pro now appears to utilise multiple processors correctly. Sure my tawdry dual-core machines don’t immediately reflect this, but cohorts with multi-core Intel machines are reporting more even distribution of processing across cores. Secondly, the application seems incredibly stable. It boots very quickly and quits cleanly. I’m also especially fond of the finer increments on faders in the mixer – no longer confined to 128 steps. Disheartening aspects include the disappearance of the ‘Samples Clipped’ message that would appear if you clipped a recording – last sighted in v7, and distributed audio processing is still a total crap-shoot. There’s also been a price rise of $100, which isn’t so surprising – v8 was pretty cheap at $649.
So am I impressed? Indeed I am. Every primary iteration of Logic is well worth the wait, and for me this version is possibly ‘the one’, or at least the one I settle on for the foreseeable future – I’d wager the computer I’m using won’t run the next iteration anyway.
NEED TO KNOW Price Logic Studio 2: $749 Upgrade from Logic Pro or Logic Studio 1: $299 Contact Apple Australia www.apple.com/au/ logicstudio Pros Flex Time functions are incredible. A more consolidated and robust application. Multiprocessor support finally works. Round trip latency times are now displayed in preferences. Guitar plug-ins add a nice touch. Cons Superior results will be with Intel Macs – forcing many to consider yet another upgrade. Not ‘supported’ on PPC machines. Summary Logic Pro 9 and Logic Studio 2 are certainly positive upgrades to a great piece of DAW software. With the Flex Time functions I’m sure Logic Pro will find an even larger user base than before. The future is looking bright for Logic Pro.
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ALTA MODA UNICOMP Just another fancy compressor or a versatile green monster? Text: Andy Stewart
I really wanted to dislike this compressor. And when I mean dislike it, I really mean I wanted to hate this thing! With so much money currently tied up in analogue compressors and other assorted outboard equipment, the Alta Moda Unicomp wasn’t something I wanted to discover a ‘need’ for. Bolting it reluctantly into the rack would therefore have to be only the first stage in a disciplined two-stage process, the second stage of which would involve unbolting it and marching it out the door before I got to know it, like it or want it. But life just doesn’t work like that sometimes… The Alta Moda Unicomp is a sophisticated, fully discrete, high voltage, complex, FET-based, Class-A, linkable dual-mono compressor featuring a capacitor-free audio signal path, feedforward and feed-back compression topologies, and RMS and peak rectifier modes. It’s a clean green compressing machine that offers truly versatile performance from a single comprehensive unit. More than capable of acting as a mastering or mix bus compressor (stereo linked or otherwise) the Unicomp can provide clear and subtle control over full-range dynamic audio, while at the same time ensuring the audio passing through the unit remains fullbodied, open and powerful. There’s no loss of energy in the bottom end as a result of shortcomings in the circuit design like many compressors on the market – it’s fast, responsive and polished in the extreme. But that’s just the beginning. COMP THAT Switching the Unicomp over to act on just one or two individual components of a mix opens you up to a world of not-so-subtle compression setups that provide you with several mix options, many of which are simply beyond the scope of the majority of compressors on the market. The Alta Moda Unicomp provides a wide-open landscape where audio can be controlled, distorted, clipped, nurtured, hammered, made thinner or fatter… all depending on what you’re trying to do within the context of a mix.
To describe the Unicomp’s various controls and how they perform, I thought it might make for a slightly less tedious read to use an example of a double bass to interrogate all the knobs, buttons and switches on its glamorous and ergonomic front panel. DOUBLE TROUBLE As many of you may have experienced, double bass can be a hard instrument to control in a mix, especially within the context of a high-energy rock track. As a close friend of mine often says about AT 28
this instrument in rock, “the best way to mix double bass is to replace it with a Fender Precision.” Given that this approach wasn’t an option, the Unicomp was patched in and quickly provided the double bass with articulation, substantial amounts of control and an entirely new attitude. The particular double bass in question had lots of dynamic range and an irregular tone… nothing new there. It was difficult to get it to sit in the mix at all, let alone have it function as the foundation of the track; one minute it was there like a ship’s fog horn, the next minute it was lost in the mist. The Unicomp was employed to provide fast acting, hard-knee compression at a ratio of 7:1, with a moderate threshold setting, a reasonably fast release and up to 10 or 12dB of gain reduction. There’s a choice on the top row of the Unicomp’s controls between ‘Peak’ mode (which is better for fast transients in many cases) and ‘RMS’ mode (arguably better for overall level reduction), as well as feed-forward or feed-back compression topologies – both of which can be chosen at the press of a nicely proportioned luminous green button [Issue 68’s On The Bench has more on how these compression topologies work]. For the double bass setup, the Unicomp was configured as a ‘feed-back’ compressor responding to ‘peak’ signals, which locked the bass down well, with virtually no pumping side-effects or collapsing of the output signal. But in the context of the mix, the bass was also fairly expressionless. In truth, it was wholly underwhelming and lacklustre. Simply compressing it a bit would have made very little difference to the overall outcome. What it was begging for was a parallel mix of some sort of distorted duplicate signal to give it some attitude and clarity. So instead of establishing a ‘mult’, as I normally would, and pushing up a distorted and compressed bass channel split via a Sansamp or similar, I engaged the Unicomp’s ‘Drive’ control. The Drive control is the first knob along the second row and is only active in the side-chain if it’s switched in with the green button immediately to its right. ‘Drive’ (formerly labelled ‘Warmth’ on earlier Unicomp models) adds second-order harmonic distortion to the gain cell of the compressor – a little or a lot depending on the amount you dial in, and in this instance I had it cranked almost flat chat. This feature also has an associated three-position highpass filter that acts upon it to drive either a full-range distortion signal through the side-chain, or one of two filtered options: the first rolling out the signal below 770Hz, the second acting above 1.3kHz. In this case I had the filter disengaged to allow the signal to be distorted across the full bandwidth of the signal. And with
the relatively high 7:1 ratio setting, the signal by this stage sounded seriously distorted, almost like a fuzz pedal.
you’re in stereo-link mode, making stereo mode as complicated as dancing the pasa doble.
The next manoeuvre involved engaging the ‘Blend’ control – again with a button – which added more of the original uncompressed bass signal directly to the output. This control also sports a high-pass filter, which again offers three positions: ‘off ’, signal passing above 280Hz or 1.1kHz. For our bass channel, this filter was switched to engage the 1.1kHz filter, providing the ability to wind untreated high-mids and tops back into the output signal. This action restored clarity to the already heavily driven and compressed signal, effectively reducing the ratio of top-end treatment relative to the bottom end. The result was a superbly controlled, articulate and vivid bass sound that was worlds apart from its mashed potato-like original. The Blend masked the heavily distorted signal surprising well and the bass sat in the mix almost miraculously, providing a low-end foundation, consistency and even melody!
COMP-LEXITY I suspect there are many engineers out there who might think that any compressor that allows you to get lost in all the settings and mangle a sound is missing the point of the device, but I disagree. Sure, a simple compressor that produces fabulous results every time is a great thing, but complex compressors shouldn’t always be misinterpreted as being ‘try-hard’ or ‘gimmicky’. Some are, but this one certainly isn’t. The Unicomp doesn’t trade away sound quality for increased versatility. I suspect these accusations are often borne out of fear anyway – some people hate devices that challenge their skill sets. Somehow the Unicomp manages to be both great sounding and highly tweakable.
OTHER SOUNDS & FEATURES The Unicomp’s versatility extends to all kinds of sounds of course, not just bass. It worked nicely on everything from mad keyboard pads and drum subgroups, to gentle vocals and string sections. Once you get the hang of the way the compressor’s various controls interact, the unit becomes fun to use, encouraging experimentation rather than inducing fear.
Admittedly, when you first approach the Unicomp you could be forgiven for mistaking it for a stereo parametric EQ, given the number of controls on its pale green faceplate. The only things that give the game away from a distance are the fantastic white gain reduction meters in the centre of the unit. It’s certainly not vintage looking as far as the knob count’s concerned. If you’re an LA-2A kind of person, the Unicomp might look a little scary at first, but I assure you it isn’t. Assuming you know how to use a compressor, the Unicomp simply offers you more control over the signal, allowing your imagination to drive the outcome rather that having to rely on a ‘classic’ design that provides no scope for tweaking. The trick with the Unicomp is to get two hands on the controls; I often worked the unit with one hand on the Blend control and another on the output. The difficulty with the unit is that adding Drive or Blend immediately increases the overall output volume, so controlling them simultaneously is the way to go… if you’re cranking in lots of Blend, for example, you’ll need to back off the output. Trickier still is the fact that none of the lower left-hand controls override their equivalents on the right-hand side when
COMP-LETING THE PICTURE There’s a lot more I could say about the Alta Moda Unicomp. It looks fantastic (apart from the two blindingly bright LEDs that blink the moment the compression detection circuit kicks in, acting more like disco strobes in low-light than mere indicators). It feels classy and well constructed (final assembly is done in Hawaii apparently, by the designer, Paul Ricchiuti). The dimpled aluminium knobs and gain reduction meters in particular give the unit an elegant individual look, while at the same time the aesthetic seems somehow timeless. The internals are beautifully constructed out of high quality components; the layout is clean, highly organised and eminently serviceable – no giant PCB boards featuring unrecognisable microscopic components here, just quality workmanship and parts. The Unicomp also has an external side-chain insert loop on rear mounted XLRs, which is armed via the front panel just below the Threshold control, and a side-chain filter that cuts in at 120 and 400Hz to allow even further manipulation of the compressor’s sensitivity to bottom end.
All in all, this compressor is one sweet ride… I know that sounds tacky but it’s true. It offers great sound performance with looks to match. I can’t really fault it, apart from perhaps wishing the filters were able to tell you where their crossover points were and wishing the LED indicators could be toned down a bit. [Actually, just as we were going to press designer, Paul Ricchiuti, emailed to inform us that the newest production run of Unicomps has mellower green LEDs]. It’s one of those compressors that feels like it might just become a secret weapon once you really got to know it. But I must resist the temptation for now… where the hell is that screwdriver?
NEED TO KNOW Price $4990 Contact Osmond Electronics (08) 8410 1111 kostas@osmondelectronics. com.au www.altamodaaudio.com Pros Awesome sound. Extremely versatile operation. A worthy tracking, mixing or mastering tool. Superb build quality inside and out. Cons Gain reduction indicator LEDs too bright. Filters have no meaningful legending. Stereo mode is complex. Summary The Unicomp is a classy compressor of the highest quality. It has high headroom, great compression control features, great looks and several tricks up its sleeve. It’s no toy or one-trick pony, nor is it just a pretty face. If you need good analogue compression, regardless of where you work, check it out.
Audio recording? From Propellerhead? It’s been a long time coming, but have the Swedes got it right? Text: Derek Johnson
Round about now, I bet many users of Propellerhead’s Reason virtual electronic music studio are wondering where their Version 5 is. After all, it’s been nearly two years since Version 4 hit the streets. It’s not here, but the Swedish chefs haven’t been away from the kitchen. Addressing perhaps the number one issue a certain type of user has with Reason, Propellerhead engineers have been busy working on an audio recording application: Record. Record – for that is the new software’s name – has the laudable aim of doing for straight multitrack recording what Reason did for virtual synths, drum machines and sample players: provide a lot for the money, and make it easy to use. Conceptually, Record has a relationship to an all-singing DAW that a home studio based around a cassette multitracker once had to a fully-fledged 24-track analogue tape machine in pre-digital days. The idea is to get the audio down quickly, and keep the computer in the background. But there is, of course, one big point: Record is software and in most of the important ways is limited only by your computer (Mac OSX or Windows XP and up) and audio hardware. If you use a quality audio interface plugged into a recent computer packed with RAM, your recordings will be as pristine and complex as you want them to be: your engineering skills and ears, not to mention the songs, will be the deciding factor of the final result. Okay, so this option isn’t as cheap as a cassette multitracker [Actually, I think it’s cheaper – Ed.], but nor is it as expensive as a do-everything DAW. And cassette multitracks don’t let you change the tempo of a song after all the audio has been recorded: Record does, and it also offers a straightforward way to manage multiple audio takes on a track, and comp the bits you like to a finished take. A QUICK LOOK AROUND Reason v5 it might not be, but Record v1 certainly has the Propellerhead ‘house style’. This is reassuring for existing Reason users, and should provide a smooth entry in for newcomers. The linear sequencer and device-laden rack offer Reason’s friendly, approachable interface, in slightly polished form. The patch cables at the back of the Rack are present, as are all the options for adding real-time or automated control.
There is one obvious innovation: Record’s so-called Main Mixer is based on the physical structure of no less than an SSL XL 9000K Super Analogue mixing desk. Linked with the ever-expandable nature of the Record Rack, this creates a flexible, powerful recording environment. Of course, Record works well with Reason although you can’t really see the connection: without the user doing anything, Reason’s devices are freely available to Record’s rack, and Reason AT 30
songs can be loaded into Record. In a way, Record is a sort of audio recording plug-in for Reason, but it turns things around and becomes the host! In any case, once you have Record on your system you’ll be unlikely to run Reason on its own again. Propellerhead’s engineers have been efficient with their coding, and on the whole your computer only has to accommodate what’s actually in the rack. The new mixer will have an impact on CPU overheads, and adding audio tracks pushes the system a bit further, but you gain so much in terms of creative opportunities. Unfortunately, some Reason users won’t be able to take advantage of this ‘upgrade’; Record will only run on Intel Macs (Mac OSX 10.4 or higher), so G4 and G5 users will have some hard thinking – or cussing – to do. Windows users meanwhile (XP and Vista) can relax: minimum spec is Intel P4/AMD Athlon XP at 2GHz or better – multiple cores (on both platforms) is recommended. And if you have a load of USB dongles attached to your music machine already, you won’t be pleased to hear that Propellerhead has also gone down that route with Record. There are options for running the software without the Ignition Key: one requires internet access for unrestricted performance, while the other lets you record and save, but not load, songs. Still, there are several users, and potential users, grumbling about this in audio forums. RACK ’EM UP Record’s Rack has inherited a number of devices from Reason so if, like me, you’ve wanted to use the RV7000 reverb device on a mix or the Scream 4 sound destruction unit on a real guitar since they arrived in Reason then you’ll be smiling. Even better, all four M-Class mastering processors (EQ, compressor, stereo imager and maximiser) have also made their classy way over to Record.
In addition, Reason’s two mixers, the DDL1 delay, CF101 chorus, and the Spider audio and CV merger/splitters are part of the arsenal. Crucial for creating mega patches of effects, the Combinator device has also migrated to Record. Surprisingly, the ReBirth input machine is here, for Windows users (the discontinued, but now free, Rebirth RB338 software doesn’t run on Mac OSX). New stuff, specific to Record, includes two amp devices licensed from Line 6, one for guitar and one for bass. The devices are each equipped with the same five Line 6 vintage amp and speaker cabinet simulations, but with a couple of small differences to make one more suitable to bass and the other more effective with guitar. And should you own a Line 6 Pod, or other of the company’s USBequipped products, the Record device can load its models, which is very handy. The addition of guitar and bass amp devices illustrates where Propellerhead is aiming its new software – bands, and musicians without a particular preference for synths and electronica. There is
…once you have Record on your system you’ll be unlikely to run Reason on its own again.
one instrument device in the Rack, though, ID8: a basic, but greatsounding, preset sound module. Its presets have been gathered from all over the Reason universe, and they’re good. Should you have call to use Standard MIDI Files, multiple ID8s will provide appropriate patches should you load an SMF into Record. The Main Mixer is linked to breakout devices in the Rack, with the link being invisible to the user. Input channels, whether for audio tracks or instrument devices, have a Combinator-like silver device loaded in the Rack. It can accommodate insert effects and provides control options, and links to other devices in the Rack.
An overview of Record, showing a small section of the huge mixer and audio doing its thing in the sequencer.
The Main Mixer’s Master Section has its own (fixed) presence in the rack, joined by the (also fixed) Hardware Interface. Like Reason’s equivalent, the Hardware interface centralises the audio ins and outs (up to 64 each) of your audio hardware, and the inputs of your MIDI hardware for customising the control of devices loaded into Record’s rack. Check out the Hardware Interface’s big meter, with pro touches such as VU, PPM and Peak reading options. SOUND TRACKS Record’s sequencer is perfect for anyone wanting quick results. There is no fuss, and focussing on two modes – arrange and edit – allows you to concentrate on the job at hand. The process is tried and true now: the Sequencer has undergone gradual evolution in Reason, and here it’s at its peak: unlimited tracks, lane-based recording, a superb comp’ing interface, and a tempo and time signature track.
Check out the side-by-side devices in the Reason 5 – err… Record – Rack!
Propellerhead has gone for a ‘clip’ vibe, though of course clips – MIDI or audio – can be the length of a song. MIDI events appear as blobs on a grid, and automation, controller and other related data are also manipulated in the same window. Cut, copy, and move tools are joined by standard manipulations such as quantisation, transposition, length and velocity offset, and so on, in a dedicated ‘Tools’ window. New for Record is a handy on-screen mouse – or keyboard-driven ‘keyboard’ window – no external controller required. Audio recording is as easy as it gets: create a track, select its hardware input (or inputs for stereo), and go. There’s a count-in, and loop recording, available, and multiple tracks can be recorded simultaneously. Audio editing is not as advanced as you might expect, but most of what you can do is non-destructive. Fades, crossfades, and clip level changes are all possible, and the comp’ing of multiple takes has a smooth ‘cut it’ and forget approach. If you find you have too much audio in your song, or don’t want the leftover bits from a comp’ing session, burn the result to its own track and erase the unwanted data.
A bit of the Main Mixer, with some sections folded out of view. The Navigator to the right previews what you can’t see.
Comp’ing multiple takes is a doddle in Record. Highlighted audio in the lower lanes make up the final take in the top lane.
The time stretching offered by Record is hands-off – you choose to have it on or off for each audio track, although you can select algorithms dedicated to general purpose or solo part use. Go half or double tempo on most material, and the audio will be surprisingly responsive. For small tempo changes, artefacts won’t be obvious at all. One important point is that although you’ll be hearing audio fully processed by whatever you have loaded into the Main Mixer, the audio is recorded direct from your audio hardware. The mixer processes audio as it plays back, not as it prints to the song file. And that’s another concept: all audio and song settings are recorded to an all-in-one file – audio files aren’t recorded to hard drive separately. Some may bemoan this Garageband/Cakewalk style of project file management ‘bundling’. IN THE MIX The Main Mixer is an infinitely expanding device, adding a channel every time you create an audio track or instrument device. The signal path is beyond comprehensive as a part of sub-$400 software: dynamics processing for each channel (and a master compressor at the end of the mixer’s signal path); eight effect send/return loops (pre or post fader); a Reason Combinator-like interface for adding whatever chain of insert effects you’d like (both on the input channels and the master section); four-band EQ with HPF and LPF filters (which can be switched into the dynamics processing AT 31
patch); and standard fader, pan, mute and solo controls. And all of it can be automated or controlled externally. Individual sections being folded out of the way make this potentially huge mixer more manageable. Also, a neat ‘Navigator’ system lets you drag your way around with miniature icon-based representations of the mixer, rack and sequencer. And to be extra flash, you can focus on any of the main windows exclusively, by ‘tearing them off ’ or using F-keys to make a selection. This is essential, since the display is largely incomprehensible if you have bits of Sequencer, Rack and Main Mixer visible at the same time.
THE MAGAZINE FOR SOUND ENGINEERS & RECORDING MUSICIANS
ALL TOGETHER NOW Record is still a ‘closed’ system, but closed means that you’re not distracted by stuff you might not use. I won’t argue for or against the lack of plug-in support, the Ignition Key security device, or whether audio should have just been added to Reason. I like the software, I like the company, and I’ll accept their choices. There’s not a lot anyone can do about it now, anyway: the software is on the shelves! But remember, Record can be a ReWire slave to any compatible ReWire host – and if you need access to VST plugs, a compatible ReWire host doesn’t actually have to be expensive (think Tracktion or Reaper).
At $399 Propellerhead’s Record is a good deal. Bands with a moderate budget wanting to record might consider buying a laptop, audio interface and Record, effectively turning their rehearsal room into an easy to use DAW-based studio. Registered Reason owners get it pretty easy: adding Record to their system costs just $249; a whole lot less than some plug-ins for other DAWs. Why would a Reason user not buy this? The audio tracks alone increase the creative potential, and the free SSL-R-Us that comes as part of that package takes mixing to a totally different level. Finally, a package bundles both programs for $899, if you haven’t yet succumbed to the Propellerhead way. Record is a great way to get your tracks down and mixed. While some of the features aren’t as deep or as involved as other DAWs, their absence makes for a no frills, easy to use and uncluttered recording (and mixing) experience. All eyes will be on the integration of third-party effects in a future update. NEED TO KNOW Price $399 Contact Music Link (03) 9765 6530 propellerhead@musiclink. com.au www.musiclink.com.au Pros Multitrack audio recording in a Reason-like environment. The Main Mixer is a virtual beast. Real-time time stretching is a session saver or creative tool. Tight integration for Reason users. Cons USB dongle. Closed to plug-ins. Modern Macs only.
Full magazine out now AT 32
The Main Mixer channel strip in all its SSL glory.
Summary The bottom line is that Record makes my Reason world complete, and I’m going to spend a lot more of my time there. A great program at a great price.
AudioTechnology THE MAGAZINE FOR SOUND ENGINEERS & RECORDING MUSICIANS
Full magazine out now
MC2 AUDIO E90
Amplifiers are getting bigger and smaller all at once. Muscle amps now rule the roost. Text: Mark Woods
Very high power amplifiers (muscle amps) are one of the more exciting developments in the ever-evolving world of professional audio. They don’t get their name from their size or weight, but rather, from their output power figures that would have been a physical impossibility only a few years ago. One of the companies leading this development is MC2 Audio, an English company involved in amplifier manufacture since the mid ’90s. One of its directors, Terry Clarke, was previously co-founder and technical director of Klark Teknik and in 2007 MC2 merged resources with XTA Audio and Quested Monitoring Systems. MC2 products have recently been made available in Australia through CMI Music & Audio. Designed for touring or installation applications, the E series amps have been available for nearly five years and the E90 is the newest and most powerful of the series. How powerful? Try 8000W into 2Ω from a 2RU amp weighing just 11.5kg. This prompts two questions: One, are the output figures real or is this another example of the type of exaggeration common in the domestic audio retail market? The ghetto blaster with the large, bright ‘800W’ sticker on the front comes to mind. And two: why do we need amps with this sort of output power? According to an independent bench test, the output figures are indeed real, and without having the capacity to test this sort of wattage myself, I’ll assume these figures are truthful and accurate. Producing that kind of output draws enough current to melt your average household electrical wiring system, so don’t bench test the amp at home whatever you do! The answer to the second question lies in the way amplifier and speaker design – along with production logistics – are becoming increasingly inter-connected. The desire for more powerful amplifiers is being fuelled by a combination of improved power handling in speaker drivers and from multiple speaker systems, particularly line arrays that can require significant amounts of power per band. Modern sub-speaker designs also rely on high power to get them moving properly and the headroom available from muscle amps allows large signal peaks to be accommodated without compromising audio quality. With the E90, line array systems using 16Ω boxes can have up to eight boxes running from one side of a single amp. Production supply companies and audio-system installers are attracted by the smaller size, lighter weight and relatively lower costs. BEEFCAKE In keeping with the muscle amp theme, the E90 looks tough and feels strong. The vented front panel is metallic blue and sticks out from the body of the amp allowing access to the removable air filters that slide out for cleaning. There’s a cutout section in the AT 34
centre enabling access to the volume knobs and power switch, as well as LED indicators for power on, audio-protect and channel link. A five-segment level/limit bar display for each channel has three green LEDs for –24dB, –12dB and –6dB, a yellow segment for –3dB and an orange segment to indicate that the limiter is working.
The vented rear panel has XLR inputs and links, a channel link button and Speakon output connectors. Each channel has a Power Reduction Circuit (PRC) facility that is controlled by two buttons used in different combinations to lower the output power level by 2, 4 or 6dB. The channel link PRC buttons have been thoughtfully recessed to prevent accidental operation and there are LEDS on the front panel to indicate the PRC is engaged. Mounting front and rear is recommended; the front rack ears are integrated, the rear mounting brackets are supplied but not fitted.
Pros Extremely high power output. High sound quality. Light weight. 2RU high.
Power comes from a Class-D design that has been engineered for efficiency and high sound quality. The switch-mode power supply maintains a large power reservoir to enable high peak power without bass sag while the power control circuits manage mains input and amplifier output currents. Built-in low-distortion limiters combined with DC, thermal and short-circuit protection keep speakers out of harm’s way, which is important since the amps may be doing the job once performed by several, so reliability is vital. The specs are very impressive: noise and THD are low, the frequency response is flat to within 0.5dB and the power figures look like they’ve had an extra zero added to them; 2150W RMS per channel into 8Ω, 4500W into 4Ω and 8000W into 2Ω! Equally impressive is their audio ability in use. The only meaningful way I could test the E90 was to connect it to the mids and subs of an installed PA that currently uses two amps for the mids and another two for the subs. In each case the E90 provided significantly more power than the two ‘normal’ amps, seemingly without raising a sweat. The mid boxes were clearer and punchier while the subs sounded like completely different boxes with more solid low frequencies, improved definition and a sense they could produce lots more level if required. The E90 is not a cheap amp but if it replaces at least four conventional amps, saves weight and size for transport and improves the sound quality, it’s certainly good value for money. That value might only be fully appreciated by suppliers and users of big sound systems but there’s also a benefit to audiences from the superior sound quality made possible by these more powerful amps. The development of amps like the E90 also allows speaker designers and manufacturers to create new products that require more power, safe in the knowledge there are amps out there that will accommodate them. Muscle amps rule, okay?
NEED TO KNOW
Contact CMI (03) 9315 2244 firstname.lastname@example.org www.cmi.com.au
Cons High current draw. Could kill your old speakers. Summary The MC2 E90 has been designed in response to a combination of speaker manufacturers making more efficient products, and suppliers and users looking for more power from smaller and lighter amplifiers. The MC2 E90 is a 2RU Class-D stereo power amplifier that weighs 11.5kg and can reliably deliver up to 8000W per channel into 2Ω while maintaining clarity and transparency. Impressive! Features Of Note Up to 8kW into 2Ω. Class-D amp technology. Switch-mode power supply for very high peak output. In-built protection processing. Five-year warranty.
AudioTechnology THE MAGAZINE FOR SOUND ENGINEERS & RECORDING MUSICIANS
SUBSCRIBE & WIN A UNIVERSAL AUDIO UAD-2 SOLO/LAPTOP The UAD-2 Solo/Laptop is a must-have upgrade for recording and mixing on your laptop. Whether you use Cubase, Nuendo, Logic, Digital Performer or ProTools, this ExpressCard-based DSP accelerator puts all of Universal Audio’s plug-ins at your fingertips, freeing up your laptop’s processor for other mixing duties. Included is the UA 1176SE compressor, Pultec EQP-1A, Realverb Pro, and CS-1 channel strip, along with a UAD $50 voucher toward plug-in purchases. One stupendously lucky subscriber will walk away with this little gem – valued at $875. Thanks the generous people at Mixmasters for this prize.
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UNIVERSAL AUDIO UAD-2 SOLO/LAPTOP & QUAD A new generation of UAD hardware is finally here, and not a moment too soon. Text: Calum Orr
Let me put things in perspective. I’ve been awaiting the release of the UAD-2 Solo/Laptop for a long time now. In November ’07 I moved from mixing and mastering music on a PC desktop machine to a MacBook Pro. At that time I figured I’d be able to happily run my four UAD-1 cards via the MacBook Pro’s ExpressCard slot with the aid of a PCI expansion chassis from Magma. Wrong! A mismatch somewhere on the great ‘firmware dating’ site in the sky put paid to that idea and I’ve been hobbling along on one – yes one – UAD-1 card ever since. I’ve also been running occasional projects back on the old PC using outdated software just to get the grunt required – a little galling but unavoidable. I’ve been using the UAD-1 plug-ins for, jeez… nearly 10 years now and with the arrival of the UAD-2 hardware relief is finally at hand. These days, literally all my mixing and mastering jobs utilise UAD plug-ins of one type or another. The company’s compressors, EQs and reverbs are all top quality. So when the new range of UAD cards was handed to me recently in a plain brown box with a nonchalant “there ya go” from Andy Stewart, I was more than eager to put them straight to work. Expectations were high and the wait had been long… tortuously long. It goes without saying that I’m a bit of a UA plug-in fan, so it was a nice surprise to discover that the newly released 5.4 software also includes review licenses for three new plug-ins: the EMT 250 digital reverb emulation, the Neve 31102 EQ and Empirical Labs’ EL7 Fatso tape simulator and dynamics emulation. Awesome! I’m totally in awe of the EMT 250 already, but more on that later. NEW SOFTWARE HARDWARE The UAD-2 Solo/Laptop is an ExpressCard/34 device designed specifically for, well… laptops. The device is about the size of a segment of Lindt chocolate attached to a credit card, if you can picture that, which gives you 21/2-times the capacity of a UAD-1.
The first thing I immediately noticed about the UAD-2 Solo/Laptop
was the improved speed and smooth running of my mastering chains. The increased processing headroom immediately alleviated the strain many of my recent sessions had suffered under, but unfortunately, projects that had previously been driven by four UAD-1 cards back on the PC still weren’t co-operating… bummer. After contemplating this persistent processing shortfall I’ve since discovered that Universal Audio has a UAD-2 Quad for laptop in the pipeline – an enticing prospect indeed. Protruding the way it does from a laptop’s ExpressCard slot, and being as rigid as it is, my only concern with the physical design of the Solo/Laptop is that moving a computer about while it’s connected might potentially lead to disaster. I have visions of placing my MacBook down on a desk, with a pen or a mic lead accidentally caught under the protruding UAD-2 card and snap! For this reason alone I think I would have preferred to see the Lindt-sized processing hub connected to the ExpressCard host via a flexible cable. Aside from this concern, it’s been smooth sailing with the Solo/Laptop. THE BIG GUNS The other UAD-2 card I’ve been using over the last month has been the UAD-2 Quad. The Quad is the new granddaddy, a powerhouse card that contains four of the new SHARC chips where the Solo has only one – or in UAD-1 terms, 10-times the power. With the Quad card running the show on my old PC, my aforementioned pre-existing Cubase 4 and Sonar 6 sessions, which had previously pushed the four UAD-1 cards to the brink, ran without a hitch. What’s more, because the sessions were playing back with ample DSP overhead, they somehow seemed less stressed and better sounding. Or maybe it was just me who was less stressed; all the memories of trying to get a mix over the line with four max’ed out UAD-1s in Logic or Cubase came flooding back, and they weren’t fond ones, believe me!
With the Quad card providing ‘an extra six UAD-1s worth of headroom’, I set about basking in the glow of the extra DSP by loading a bunch of new plugs from more recent UA software releases. Having this extra system headroom was a relief and a real bonus. With some re-energised mixes uploaded to past clients as a valueadded remix favour, I then set about mixing completely new material, again using the Quad via my PC setup. As a test of its power I decided to use nothing but UAD plug-ins for an entire mix. The first of these wound up running 48 mono plug-ins, with a further 12 stereo plugs on some stereo signals like overheads, acoustics, effects returns and buses. Of course, heaps of DSP is never enough and it goes without saying that I eventually maxed out the single Quad card. Hey, I wouldn’t be doing my job if I hadn’t. MAD MAX-2 In the same way as you could with the UAD-1 card, it’s easy enough to add more UAD-2s to a system provided your wallet can cope. In fact, a single system can now run up to four UAD-2 and four UAD-1 cards. This represents a serious amount of grunt – 44
times the power of a single UAD-1 by my calculations. However, at a little over $2500 for a single Quad, a system of this size will set you back a pretty penny. Ten grand, to be exact! To this end, I reckon a UAD-2 ‘Deca’ should be the next release on the cards (if you’ll pardon the unfortunate pun). This would give the next release from UA 25 times the power of a UAD-1 on a single card and 100 times the power if you had four! This may seem like a throwaway and churlish comment, but already there are some voices criticising Universal Audio’s decision to use the SHARC chip, when other chips were available that could have provided the plug-ins with many times the SHARC chip’s capabilities. THREE NEW PLUG-INS Speaking of plug-ins, new to the ever-expanding UAD software emulation empire are three new plugs in version 5.4. First cab off the rank is the EMT 250 digital reverb, which is, in a word, superb. From the moment I clapped ears on it I began to wonder if I hadn’t heard its sonic signature before on records like Beck’s Seachange and Mutations – the EMT 250 hardware unit at the source of this emulation is, after all, housed at Ocean Way where Beck recorded and mixed these albums with acclaimed Radiohead producer, Nigel Godrich. Although I suspect there’s probably also a judicious use of Lexicon 480L on these records, after using the EMT 250 emulation myself, I don’t believe Mr Godrich could have passed up the opportunity to use the original!
Just to fill you in on some history, Ocean Way Studios and Universal Audio have a very long association that began with Bill Putnam Senior owning the original studio and premises that now contain Ocean Way. These days Bill Putnam Jr and Ocean Way owner, Allen Sides, collaborate on software emulations of hardware classics such as the beloved Fairchild 670 and now the EMT 250. From the moment I patched in the EMT 250 it was obvious this was a classic sounding reverb. Deep, interesting and never hohum, the algorithms developed way back in 1976 by Dr Barry Blesser still sound great today. I can always find something in a mix that can do with a healthy dose of EMT, and between this and the UA Plate 140 and Dreamverb, I really don’t need any other reverbs (though I still use the excellent Platinum reverb and Space Designer Convolution reverb built into Logic Pro). Special mention must also go to the EMT 250’s graphic display, which does a great job of emulating the appearance and mechanical workings of the original unit. Virtual arc faders like the ones found on ’60s EMI consoles provide delay, pre-delay and low and high-pass filter control with five buttons for engaging different effects such as Chorus, Delay, Phase, Echo and Space. The 250 looks very ‘schwing’ but some may argue that it takes up too much screen real estate. NEVE 31102 EQ The new Neve 31102 EQ plug-in and its ‘lite’ cohort, the 31102SE, are also welcome additions to UA’s already impressive Neve emulation suite. Modelled on Class-A/B architecture from the
late ’70s, the 31102 has a flavour all its own. It has a nice hard sound when you push the boost over +9dB, plenty of solid tone and a less pronounced top-end sheen thanks to the frequency selections and topologies, which differ from the more familiar Neve 10-Series EQs. I’ve used the real version of this EQ in Dave Clooney’s Atlantis studios in Port Melbourne where I’ve previously had great results on bass, snare and vocals. I didn’t manage to get the plug-in and the original hardware side by side, unfortunately, but the UAD plug-in certainly operates and sounds similar to my memory of the original. Next to bass and vocals, electric guitars have received the most 31102 processing. The EQ delivers grunt but never at the expense of clarity – perfect for electric guitars. The 31102 also has some serious attitude between 100 and 300Hz without generating mud. I also like the way you can get rid of top-end wispiness without affecting the midrange sheen. All in all it’s a very useful EQ that’s optimised nicely for the UAD-2 Quad, allowing you to get quite a lot of instances. The Crème de la crème and probably the most anticipated of the three new plug-ins is the Empirical Labs EL7 Fatso Jr/Sr. Many users have had tape emulation on their UA wish-list for a long time now, and UA has finally come through with the goods, producing a faithful rendition of Dave Derr’s modern classic. I’ve been strapping the ‘Senior’ version of this plug-in over the output bus of several of my recent mixes and I can happily report that it has sounded beautiful on almost every mix. It’s also been great on individual mix elements like weedy acoustic guitars and bland backing vocals – improving their tone and attitude significantly. NO NUMBER CRUNCHING I’m going to resist the temptation here of getting caught up in descriptions of how many instances of which plug-ins can be achieved with the three new UAD-2 card options; Universal Audio has that information on its website if you’re interested. I will point out, however, that with the advent of the UAD-2 a couple of negatives have crept into calculations. Firstly, for licensing reasons only UA understands, the Nigel guitar simulator and all of its plug-in modules have been discontinued for use on the UAD2 platform. Secondly, it seems many of the new plug-ins will no longer run on the UAD-1. Already the Cooper Time Cube, EL7 Fatso Jr and EMT 250 are restricted to the UAD-2 platform only.
NEED TO KNOW Price UAD-2 Solo/Laptop: $875 UAD-2 Solo: $875 UAD-2 Quad: $2561 Contact Mixmasters (08) 7200 4500 firstname.lastname@example.org www.mixmasters.com.au Pros Quad card impressive and fast. New plug-in emulations of v5.4 worthy additions. The software suite is the best of its kind. Cons UAD-2 Solo/Laptop still no powerhouse. Solo/Laptop is potentially a fragile device. Summary The UAD-2, in all its forms, have been a long time coming. They’re an impressive group of hardware-for-software tools, which, like the previous incarnation, provide superb emulations of fabulous equipment. I’m not sure their boost in DSP is quite the leap many had anticipated but in clusters of two or three, the Quad card in particular offers some serious processing power.
It’s a fact of life I suppose that I’m now on a UAD hardware upgrade treadmill. But since UA plug-ins are some of the finest going around I’m happy to bite the bullet. Would I have preferred a larger boost in power from the UAD-1 to UAD-2 Solo? Yes. For now though, I’m quite content pursuing my new threshold of ‘maxed-outness’. With the Quad card in particular, you’re a lot further into a mix before you start having to go down the ‘bounce plug-in to disk’ route. Call me weird, but such is my commitment to the UAD platform I’m now aiming to purchase an 8-Core MacPro running two UAD-2 Quads. For us ol’-timer UAD1 users, that’s the equivalent of 20 UAD-1s. Some may snicker, others will marvel.
HOW YOU GET THAT CLASSIC TONE …tubes are optional. Text: Rick O’Neil
You would think I’d know better by now. I’ve had a debilitating migraine for a couple of days but now that it’s finally subsided and my wife and kids have gone to sleep, I’ve nicked downstairs to my workshop to try and get some semblance of a magazine article together while I’m still feeling human. But when I first got down here I figured ‘what the hell, I’m not gonna sit down here and type on the computer! I’m feeling good, my focus is strong and I can see past the end of the table for the first time in days’. So I looked around my shed for a distraction – and believe me there are plenty of those around here – and what did I find? My mini sign-writer’s airpowered spray gun, and before I knew it I’d mixed up some paint and given those old tube microphones I’ve been restoring a new coat. A word of advice here: if you’re feeling good precisely because you don’t have a migraine, and if you think that breaking out your chemistry set to mix up a concoction of lacquers, thinners and toners to recreate the exact shade of metallic golden silver that Neumann once used in the ’60s is going to leave you feeling good, think again! But by the time I’d realised this I’d already mixed the brew, so I ran a couple of coats over the parts anyway – once you set up a spray gun and get the paint mixed, as the saying goes, “You’re committed to the deal.” Feeling a bit woozy from the head full of thinners, I then sat down to watch the paint dry… exciting stuff. ‘Oh great,’ I thought as the entertainment levels hit new heights, ‘No doubt this migraine will comes back with a vengeance after messing with all those chemicals. Oh well, I’d better clean up and do something else.’ I decided it was time to flick through my box of badges and make a decision about which ones were going on which microphones. I assume, dear reader, that you too have a box of Neumann badges under the AT 38
house and a selection of broken microphones waiting for that night when you’re in the mood to tackle a restoration job, right? Of course you do, otherwise you might think I’m a bit strange, and we can’t have any of that! Anyway, I picked out two badges that are the right vintage and size and spent 10 minutes pretending I was about to stick them onto the microphones, checking and rechecking the rivet holes for alignment. I’m not happy with the line up, but of course I’m now distracted by the discovery that the badges also need restoration before I can put them on the mics. I know I’ll never paint them after the fact because it’s such a bitch of a job, so I’ve decided I might as well touch up the badges as well while I still have the paints out. But before I can stop to remind myself about the looming reemergence of the migraine, I crack open the little pot of black nitrocellulose lacquer that I keep for just such an occasion. You’ve all got one of these tins under your house too, right? It’s the most useful paint in the world – dries in seconds and smells… err… a bit like the stuff they tried to stop you sniffing when you were 15. You know the smell; we all know the smell. It smells like a great way to re-launch a migraine. THROUGH THE LOOKING GLASS So here I am looking at some tube mics in a dozen pieces, watching paint dry, waiting for the migraine to hit, and writing this article. I’d better get this written I guess while I can still think straight so here goes… if it reads weirdly, I blame the nitrocellulose. [I presume this mean you sniff that stuff every issue then – Ed.]
I’ve just picked up one of the microphones I’ve been working on and stopped for a minute to marvel at it – its triumph of engineering. In the grand scheme of things microphones are quite simple devices and of all the types ever developed, valve mics are some of the simplest. The actual valve (or tube – it’s up to you which word you use) is usually just an impedance
changing device, which buffers the capsule down to something more usable – capsules have an incredibly high impedance. Sometimes these valves add some gain, but often not too much. That’s the thing with most tube mics, they don’t need to have complicated gain circuits in them, and as we all know, in audio, less is more. There are some more complicated tube microphones, like the Neumann U67, but these are the exception rather than the rule in the tube mic line-up, and I’m talking about old tube designs for the moment here; I’ll get into new designs in a minute. I am pretty sure I know enough about audio tube electronics to be a danger to myself but not quite enough to be dead just yet. I must admit I’m scared of tube electronics – it’s a good way to be actually; the voltages involved can easily kill you dead! And let’s face it; nobody wants to wake up dead! Because of this danger, lots of people much smarter than myself have decided not to play around with tubes, but for good or ill, I’m not lots of people. DISTORTING THE FACTS I know I’m going to get into trouble with this article, so I might as well just dive straight in and let the cat out of the bag.
In my estimation 95 percent of the pro audio community simply has no idea what they’re talking about when it comes to vacuum tubes; what they do and, more importantly, what they don’t do. Let me start with a doozy (ahh it must be the thinners): vacuum tubes, valves, tubes or whatever you want to call them simply do not equal distortion! In fact, vacuum tubes, and especially triodes, continue to be the lowest distortion amplifying elements ever made. No germanium or silicon transistor, JFET or MOSFET has ever approached the distortion performance of the direct-heated triodes, with indirect-heated triodes following closely behind.
So you understand what I’m saying? This is my point again in case you missed it: tubes are the cleanest analogue single gain stage in the world! ‘But Rick, what about all that stuff we all know to be true about tubes having that pleasant second-order harmonic distortion? What about the fuzz, what about the warmth? What about the noise tubes make? Come on, we all know tubes are downright dirty, fuzzy mothers. So what gives?’ Okay, well let’s clear some stuff up here. Real pro audio tube circuits run on high voltages, have massive headroom before the onset of distortion and are well and truly quiet enough for you to make a recording of any band you care to nominate. But what’s also true is that most of the classic tube equipment you find in a studio or under somebody’s house is really old and probably has a host of electrical faults. I can almost guarantee you most of the vintage valve gear you use doesn’t sound or perform the way it was originally intended, or the way it could if it were serviced properly. But let’s forget that for the moment – if the valve gear you’re referencing in your mind isn’t up to the design specs it was originally built to, we shouldn’t include that in our thinking here. The reality is, if it was ever designed to work in a recording studio, with the exception of a couple of poorly designed oddities, the classic tube pro audio device was not designed (nor even inclined to) distort in normal use. It’s the new-school modern tube stuff that distorts, not the old stuff. PULL THE OTHER ONE Let’s take, for example, the king of the studio tube equaliser, the venerable Pultec.
Well guess what, you cannot make a Pultec distort by turning up all the knobs no matter how hard you try! Despite what the plug-in does or what your mate’s audio interweb school brain has taught you, you might distort the next thing in the chain (probably not a tube, by the way), but you won’t distort the Pultec.
It is, for the most part, a myth – the product of misinformed people reinterpreting and regurgitating Chinese whispers until the untrue becomes the unquestionable.
‘And what about how noisy those Pultecs are?’ Actually I use restored Pultecs in my mastering suite every day and their respective noise floors are quieter than that of my Manley Massive Passive or Vari Mu. In fact, they’re quieter than a bunch of solid-state stuff I have as well. Sure the tube stuff is not the quietest stuff in my setup, but it’s all useable. In fact it’s very useable and doesn’t distort in any way that could be labelled euphonic or brimming with second-order harmonic distortion. Are you one of those people who says those words in public? I hate phrases like ‘euphonic colouration’ and ‘second-order harmonic’. I propose a two-month ban on them across all forms of print media and the internet. Who’s with me? … Crickets chirping in the background… ‘Okay, okay, we know that Rick, but tubes are still ‘warm’ right?’ We all know that!’ Well frankly, no, we don’t all know that. It’s not actually the tube that’s warm sounding – actually, let me rephrase that: it used not to be the tube that’s ‘warm’ sounding. Nowadays there are so many shitty tube harmonic distortion boxes designed for the studio that supposedly enable you to get back some of that ‘vintage warmth and distortion’ that I almost didn’t bother writing this article. If I didn’t know better I would say it’s a lost cause by now... In the search for ‘warmth’ everybody hits the poor old tube right on the head, but the tube itself was originally designed to be clean and offer DC to light frequency specs. It’s the non-linear transfer characteristics of other stuff in a box that we need to look at, but let’s hold that thought for a moment while I try and explain how we all got to the point where we’ve rewritten history and decided tubes are ‘grungy’. OVERLOAD OF RUBBISH The whole Chinese whisper about tubes and their distortion characteristic comes from the tone of guitar amps, or more specifically, the power section of a tube guitar amp. If you overload a guitar amp’s output section, there’s a fair chance you’re going to hear the second-order harmonic distortion that we all know and love as being ‘tube distortion.’ That’s where the sound is, that’s where the analogy comes from, that’s the kind of distortion we like.
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This line of self-perpetuating misinformation has been around for so long now that the poor old vacuum tube has just about run out of ways to wipe the mud off its name.
It’s actually a real pro device with real headroom.
What this simply means is that when I think tubes I think supremely clean, perfect amplifiers. When most other people think tubes they immediately think ‘warmth’, ‘euphonic colouration’ or plain old dirty ‘saturation’.
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Most old tube hi-fi amps run the same kinds of power tubes as your normal guitar amp: the EL34 and 84 or the 6L6 – the 6L6, by the way, has been in continuous manufacture since 1932; a record unmatched by any other electronic device! There must be something going on there.
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The thing is, hi-fi amps are rarely if ever designed to distort (or soft-clip – whatever you want to call the ‘tube thing’). So outside of a guitar power amp clipping, where is this tube saturation we all talk about, read and covet so much?
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It must be in the tube mics, right? Well no, not really. Sure there’s a distortion element to a Neumann U47, for instance, but is it actually in the tube or is it in the core saturation of the transformer caused by the small amount of gain that the impedance vacuum tube circuit sets up? Bingo! You want to talk about the small amount of distortion in a few vintage tube mics, let’s talk about transformers, but even then, compared to the screaming tube guitar sound it’s a non-starter – let’s look elsewhere. ‘What about the tube mic pre then, Rick, they saturate right?’ Err well no. The new ones do sure, but that’s because everybody’s been talking about tube colouration for so long that an entire generation of tube designers have retired hurt and the guys left playing with the circuits think it’s their job to put the distortion in ‘just like the good old days’. So just where was that distortion coming from in the good old days? I think it was in the same old place it always was: in the guitar amps, the Leslie cabinet, the Farfisa organ, the Wurly. It was coming out of the low-headroom transistor desks that have been around since the mid ’60s and the tape-recorders that suffered at the hands of engineers that should have known better than to record a great take at the wrong level! Now all this is not to say that distortion is bad – hell no, distortion is a great tool. In fact, I believe we need it to make music in the modern context. I can think of a hundred ways to fuzz something up to make it sit better in a track. The point I’m simply making is that in the scheme of things, tube colouration from classic pro audio gear is, in fact, not tube colouration at all. It is, for the most part, a myth – the product of misinformed people reinterpreting and regurgitating Chinese whispers until the untrue becomes the unquestionable. Actually before we get too far down the line let’s just clear something up before I get hanged by every SAE student with a rope. It’s unlikely that your classic tube mic or your classic tube mic preamplifier is distorting in a way that even slightly resembles the sound of a guitar power amp section distorting. The mini tubes in a mic just don’t have the input level to clip – unless you put a cannon in front of it, at which point the capsule would be wiped out anyway. The classic mic pre – as in a Telefunken V72/76 – does not have the gain structure to allow it to clip internally, so let’s just rule out the ‘front end’ second-harmonic soft overdrive/saturation of the classics shall we. What has happened in the last decade or so is that all the new products have come out with features that never actually existed in the vintage suspects – the ‘classic’ if you will. We’ve simply created a myth and invented its rebirth. But as I said, don’t get me wrong, you can really use the distortion devices that the
studio world is now full of. But if you’re looking for that ‘classic tube sound’ as it’s described and prescribed in every audio journal, website and audio school the world over, you must understand it for what it is – the modern incarnation of a misunderstanding. ‘But Rick, my old tube mic is still magic, right?’ Well, sure it is. I wouldn’t bother sniffing paint fumes all night to get mine looking and working like new if they weren’t. But that ‘tube sound’ – as we all inaccurately describe it – actually lurks in other components that are a bit harder to see behind those spectacular glowing tubes. TRANSFORMING OUR PERCEPTION We’ve all learnt about transformer sounds by now. “The sounds is in the iron” is a phrase I hear repeated almost daily, and it certainly is, but I know the guy who coined that phrase personally and he was talking specifically about Neves. But I suppose it applies to anything really, lots of things have a sound; lots of things have the inability to pass a square wave. But I think you’ll find the biggest culprit creating distortion is the human being in the chain – by default or design we just seem to like things that aren’t squeaky clean.
I was going to waste three paragraphs here talking about tube compressors and their sound, but in a world where there are now 5000 modern replacements for the five commonly used tube compressor brands, all with in-built guitar amp simulations to recreate the classic ‘drive’ of the original, it hardly seems worth it. ‘Aha! But Rick, surely tube compressors distort?’ Well you’ve got me there! Some of the classic tube limiters like an LA-2A or a Fairchild 670 can certainly distort but the thing is, most of the time long before they ‘clip’ they’ve sucked out all of the top-end tone from the sound. And in their defense this is with the knobs working way out of their usable range. A classic solid state 1176 by comparison wants to distort on the test bench while they’re still building the damn thing – so much for tubes being the dirty ones. THE CLASSIC TONE So what’s the fuss all about, Rick? You say you like some of the new fuzz-boxes, you understand why people want to make things distorted in the mix, and you know how much fun it can be, so why the lengthy diatribe?
Well, I don’t know really… I guess it’s just that the story of guitar amp distortion has become the story about classic tube pro-audio, and it’s wrong, wrong, wrong. So if we’re to be honest about that ‘tube sound’ we all think is in our mics and preamps (but isn’t really) let’s just call it ‘the classic tone’ from now on shall we? And now that we understand all that, what about replacing all of those crappy $99 low-voltage ‘tube sound’ pro audio wannabes with something that actually does sound like the classics? Here is how you do it if you’re up for a challenge. Start with a simple elegant electronic design, put it inside some kind of stylish case that can be dropped more than once, make sure it sounds great when you make music with it and give it buckets of headroom. Put controls on it that are limited to what the thing does best so that anybody can recognise its individual footprint instantly and make sure – just like the microphones I’m playing with tonight –that it’s still repairable in 50 year’s time. In my opinion, that’s how you get a classic tone – tubes are optional. Rick O’Neil runs Turtlerock Mastering in Camperdown – he has lots of tube stuff.
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