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JULY 2016 TM


Classic Tracks: ABC Trevor Horn on The Lexicon Of Love

Tom Oberheim Two-Voice Pro

An analogue classic is reborn

John Carpenter Cult director, composer and pioneer of electronic music

NEW! Modular synth column inside





WORTH $5900




THE WORLD’S GONE MODULAR In the May issue of SOS I compared guitarists and keyboard players and their respective tendencies to modify their instruments. While keyboard players mostly seem happy with what comes out of the box, guitar players don’t seem to be able to resist experimenting with changing pickups and other hardware. The obvious exception to this rule is the modular synthesizer, and one of the more surprising directions that the synth market has taken recently is the huge resurgence of interest in all things modular. We have established makers such as Doepfer and Moog being joined by an ever expanding army of boutique builders, many subscribing to the Eurorack format. At any music show where modular synths are on display, there always seems to be a crowd around them. I say this renewed interest is surprising only insomuch as the programmable synthesizer was developed in response to the demand by keyboard players for an instrument that offered rapid access to different sounds. Now it seems we’ve come full circle, and gone beyond the ‘it must be analogue with knobs’ stage to a fascination with patchable modular instruments. It could be argued that if you buy

all your modules from the same manufacturer, then the range of results you can achieve will be no different from a digitally programmable analogue instrument that incorporates the same oscillators, filters, modulators and envelope generators along with a flexible routing and modulation matrix. However,

“At any music show where modular synths are on display, there always seems to be a crowd around them.” Paul White

Editor In Chief

July 2016 / w w w . s o u n d o n s o u n d . c o m


the beauty of discrete modules is that you can mix and match them, and as long as they subscribe to the same control voltage and trigger protocol (or there are converters available), you can create sounds that are not available from any single off-the-shelf instrument. And if you need more oscillators or filters, you can add more modules. The only trouble is that there are now so many modules available that the choice can seem overwhelming, and that’s why we’re launching a new column devoted to modular synths in this month’s issue. Of course we’ve been reviewing modular systems in Sound On Sound since the 1990s, but it seems like the

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time is right to embark on a regular article dedicated to reviews of individual modules, and hopefully help modular synthesists — and would-be modular synthesists — make sense of the vast array of gear now on offer. And while we’re on the subject of synths, this might be a good time to mention that we’re launching our own festival of synthesizers at this year’s Sensoria Festival in Sheffield. Called, appropriately enough, SynthFest UK, the event will take place on Saturday 1st October, and if you’ve the slightest interest in synths it should be a key date in your diary. Check out for more information and tickets.



Production Manager Michael Groves Designers George Hart, Alan Edwards & Andy Baldwin Classified Production Michael Groves

Editorial Editorial Director Dave Lockwood Editor In Chief Paul White Technical Editor Hugh Robjohns Features Editor Sam Inglis Reviews Editor David Glasper Reviews Editor Matt Houghton Reviews Editor Chris Korff Production Editor Nell Glasper News Editor Will Betts


Chairman Ian Gilby Managing Director Dave Lockwood Finance Manager Keith Werthmann Administration Assistant Nicole McCammon

July 2016 / w w w . s o u n d o n s o u n d . c o m

Circulation Manager Luci Moore Circulation Administrator Lisa Pope


New Media Director Paul Gilby Design Andy Baldwin Web Editor Ellis Sutehall

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July 2 0 16 / i s s ue 9 / volum e 31 w w w. s o u n d o n s o u n d .c o m


FEATURES Sylvia Massy: Adventure Recording A unique insight into the unusual production techniques of engineer, producer and author Sylvia Massy.

Mix Rescue Our engineer reminds us that you don’t have to cram every microphone signal into your mix just because it’s available.


Recording Resonator Guitars Part six‑string, part speaker cabinet, resonator guitars are like no other instruments. Here’s how to capture their distinctive sound.

Classic Tracks: ABC 1982’s ‘The Look Of Love’ paired an ambitious band with an ambitious producer, and the result was a perfect piece of pop music.

Inside Track: Deftones The success of Deftones’ Gore is

reward for the band’s restless experimentation — and for the work of engineer, producer and mixer Matt Hyde.

John Carpenter: Film Director & Composer John Carpenter is not only a cult movie-maker, but also a pioneering electronic composer. John, his son Cody and collaborators Alan Howarth and Daniel Davies explain how it’s all possible.


Studio File We travel to the sunny banks of the river Mersey to visit Liverpool’s Motor Museum studios.

Notes From The Deadline Sometimes a make-or-break opportunity can make you and break you...

The Mix Review Our mix engineer analyses tracks by Zayn, Jonas Blue, Alan Walker, Kendrick Lamar and David Bowie.




WORTH $5900


Pioneer RM07 Nearfield Active Monitors  om Oberheim Two Voice Pro T Analogue Synthesizer Keith McMillen Instruments K-Mix Digital Mixer, Audio Interface & DAW Controller Positive Grid Pro Series Compressor Modelling Plug-ins DMG Audio Limitless Mastering Limiter Plug-in IGS Audio Springtime Four-channel Spring Reverb Phaedrus Audio Phamulus Mono Vari-mu Compressor

WORKSHOPS We learn how to exchange tracks with other musicians in Pro Tools. Tighten up your timing without sounding robotic using Logic’s quantisation features. Streamline your use of Sonar’s automation with these time-saving tips. We explore some useful functionality in Live’s Arrangement page. Studio One offers the full gamut of automation features — but which ones are appropriate to your needs?

Studio Electronics Quadnic Eurorack Digital Oscillator Module AJH Synth MiniMod Transistor Ladder Filter Eurorack Filter Module Mutable Instruments Rings Eurorack Resonance Module Triad Orbit Mic-stand System Behringer X-Touch & X-Touch Compact Control Surfaces Polyverse Music I Wish Granular Synthesis Plug-in Waves Abbey Road Reverb Plates Reverb Plug-in Eventide T‑Verb Reverb Plug-in PSP B‑Scanner Vibrato & Chorus Plug-in Elysia Karacter 500 Analogue Saturation Processor Hypersynth Xenophone Analogue Synthesizer Acustica Audio White Equaliser Plug-in APS Klasik Active Nearfield Monitors Moog Model 15 App Synthesizer App For iOS Apple 9.7‑Inch iPad Pro Tablet Computer IK Multimedia iRig Pro Duo Audio Interface Audio-Technica BP40 Dynamic Vocal Microphone Aston Microphones Halo Portable Vocal Booth Sonic Farm Silk Road Microphone Preamplifier Waves G  reg Wells Plug-ins Mix Bus, Piano & Voice Processing Plug-ins Bright Sparks Film & Album Hamstead Soundworks

With Cubase, you can implement an innovative parallel compression technique first conceived by Andrew Scheps.

Tom Oberheim hasn’t just revisited his classic Two Voice synthesizer, he’s made it even better.

Signature Analogue Tremolo Guitar Effects Pedal Electro-Harmonix Mel9 Mellotron Emulator For Guitar Samson Z55 Closed-back Headphones Mojotone Quiet Coil Noiseless Guitar Pickups LIVE Fender Fortis F-128BT Active PA Speaker LIVE Alto Mixpack 10 Powered Mixer & PA Speakers LIVE Turbosound iP1000 Column PA System Sample Libraries Gothic Instruments Dronar Hybrid Module Xtant Audio Model Brass Wideblue Sound Eclipse Big Fish Audio Country Guitars SOUND



Behringer’s X‑Touches offer affordable DAW control and tight integration with the company’s X‑Air system.

REGULARS News Q&A Off The Record Showcase Live News Sounding Off

July 2016 / w w w . s o u n d o n s o u n d . c o m

N EWS W W W . S O U N D O N S O U N D . C O M / N E W S

This year’s model

Moog remake Minimoog Model D and launch Modular iOS app


oog Music’s annual Moogfest event has grown exponentially in recent years, developing into a combined music festival, technology conference and four-day celebration of all things synth. The talk of the 2016 event was the return of the Minimoog Model D. After more than 30 years, this most iconic of synths was officially back in production, albeit a pilot-production run handcrafted in the Moog Pop-Up Factory specially for the festival. However, given that the clamour for the Werkstatt-01 self-build synth produced for Moogfest 2014 meant that the kit was soon officially released, we can feel very positive indeed about seeing a full-production Model D at some point in the near future. As for the limited-run Minimoogs built for the festival — which were only available to purchase at the event, costing a reported $3499 each — they were an almost exact recreation of the original Model D, the only difference being the inclusion of MIDI for note control. The engineers at Moog learned a lot about recreating their own historic products while making the Model 15, System 35 and System 55 at the beginning of 2015, and have transposed this knowledge into the rebuilding of their most famous design. Through-hole electronics are used throughout, meaning that the old-spec components could once again be employed. Moog even went so far as to commission special runs of obsolete components from external suppliers in order to get the most authentic sound. We’ll be bringing you a full and detailed examination of the limited-run Moogfest Minimoog Model D very soon. Meanwhile, we wait on tenterhooks for further news of a full-production version.

In other news, Moog Music have launched a new app for iOS. The Model 15 app is based on the hardware modular system of the same name. It’s compatible with all 64-bit iOS devices and guides musicians through the process of using this formidable synth. Moog have tried to use the possibilities of a touchscreen display to the utmost, allowing users to zoom and pan around their modular patches with ease. Both monophonic and four-voice polyphonic modes are included and, in addition to support for an external MIDI keyboard or controller, the Model 15 app provides a range of on-screen control options, including a traditional keyboard with pitch and mod wheels, a ribbon controller, an Animoog-style keyboard and an eight-step sequencing arpeggiator. The Model 15 app is available to download now from the App Store, priced $29.99. Moog Music Inc +1 828 251 0090

Hammer time

Hammer Audio debut with large-diaphragm capacitor microphone


nnouncing their intentions with a slogan that proudly proclaims, “Made by sound

cardioid, omnidirectional and figure-of-eight polar patterns, and switchable low-cut filter that can be set to turn over at 80

engineers for sound engineers,” Hammer Audio are a new British company dedicated to high-quality, handmade audio products. The first blow to be struck is the HA-872, a large-diaphragm, multi-pattern FET capacitor mic that features a Neumann U87-style isolated dual-backplate capsule. Entirely hand-built, right down to the plating on the body (which is said to give each individual mic its own unique look), the HA-872 would appear to be designed for flexibility as well as quality. Hammer Audio say that the mic has a neutral, balanced character, an excellent off-axis response and a huge dynamic range, as well as a maximum SPL quoted at 142dB at 1kHz — but that’s not all. In addition to the choice of

or 180 Hz, the HA-872 features two distinct voicings. In ‘vintage’ mode, the mic’s frequency response is said to imitate the classic FET microphones of the 1970s, with a significantly reduced high-frequency peak. Switch to ‘modern’ mode, however, and this de-emphasis is removed and the mic reverts to the more open and airy sound typically of a contemporary capacitor mic. The HA-872 comes in a custom flight case, accompanied by a bespoke shockmount and a high-quality three-metre cable, and will also be available in matched pairs. Expected to start shipping in the first quarter of 2017, the HA-872 is available to pre-order now, with US pricing yet to be confirmed.

Pleasing squeeze

Empirical Labs get turned onto digital with Arousor plug-in


ith its large, cream-coloured control knobs and pleasingly aggressive yet versatile sound, the Empirical Labs Distressor is one of the most easily identifiable and widely used outboard compressors around. Now over 20 years old, it’s a bona fide modern classic, but Empirical have resisted the clamour to follow many of their peers and create a plug-in version — until now. But instead of producing a direct port of this celebrated bit of hardware, the much anticipated Arousor plug-in goes beyond the remit of the original, adding a host of new controls while promising to deliver the Distressor’s classic ‘knee compression’ inside the box. Alongside those familiar white knobs, the Arousor features a new attack modification section, which lets the user

attack. In place of the Distressor’s three distinct harmonic distortion modes, there’s a soft clipping section with a continuously adjustable saturation control. Similarly, there’s a detector high-pass filter control and a fully featured detector side-chain EQ, allowing you to set the frequencies you want the compressor to ignore, emphasise or suppress. A dry/wet blend control for parallel compression is also included, as well as a wider range of ratio settings. The Distressor’s brick-wall ‘Nuke’ setting has been replaced by ‘Rivet’, designed for levelling out peaks and flattening room mics. Emprical Labs’ sense of humour is also evident in the clip warning light, which is additionally labelled ‘BAD!’ This is perhaps an indication of the designers’ love for analogue processing and related

adjust the curve of the compressor’s

distrust of digital, but Empirical say great efforts have been made to minimise digital artifacts and preserve non-linear processing, while keeping the CPU load within reason. They also hint that further development of the plug-in’s features, user interface and under-the-hood processing are to come. The Arousor plug-in is out now for Windows and Mac OS in AAX, VST2, VST3 and AU formats. It requires iLok 2 and costs $349 direct from the Empirical Labs web site. Wave Distribution +1 973 728 2425

Take five

Arturia V Collection 5 adds five new classic keyboards


he latest version of Arturia’s celebrated V Collection bundle of modelled keyboards is out now. It features five new additions, taking the total to 17 and rendering this assembly of vintage and modern synthesizers, organs and electric pianos even more comprehensive. Employing Arturia’s TAE (True Analogue Emulation) algorithmic modelling, V Collection 5 promises authentic, detailed sounds with an expanded set of features not found on the originals, from polyphony to additional automatable controls. Of the five new virtual instruments in this fifth instalment, perhaps the most interesting is Synclavier V, an exacting emulation of the Synclavier created in collaboration with Cameron James, who created the software for the original. The first commercially available digital synthesizer, the Synclavier combined FM and additive synthesis and was one of the most sought-after — and unobtainably

strings, hammers and soundboard. Alongside these new entrants, the Arturia’s 12 existing virtual instruments are all present, from emulations of the Vox Continental Organ and Wurlitzer piano to classic synths like the ARP 2600, Yamaha CS-80, Roland Jupiter 8, Oberheim Matrix 12, Sequential Prophet 5, Moog

expensive — synths of the 1980s. Based on the original source code and featuring many of the original presets, Synclavier V goes several steps further, adding a host of new synthesis possibilities. Elsewhere, V Collection 5 also adds the B-3 V tonewheel organ emulation, the Farfisa V virtual Farfisa Compact Deluxe organ, the Stage-73 V Fender Rhodes-style electric piano emulation, and Piano V, a collection of nine different acoustic pianos ranging from concert grands to studio uprights. Fully modelled, Piano V lets you adjust a whole host of physical and mechanical parameters, from mic position to the

Modular and Minimoog. All come with new, Retina- and 4K-compatible resizable GUIs. The bundle also includes Arturia’s Analog Lab 2 software, providing a single environment from which to access the entire collection of sounds and presets. V Collection 5 is also compatible with Native Instruments’ NKS control standard, while MIDI Learn functionality should make it easy to map parameters to other MIDI controllers. Operating in AU, AAX, VST2 and VST3 plug-in mode or as a stand-alone instrument, this impressive collection is out now priced $499.

Wunder toys

Wunder Audio announce new Suprema Line mic range


onsole, outboard and mic manufacturers Wunder Audio are revamping their entire microphone catalogue. Based in Austin, Texas, the company produce a comprehensive range of high-quality mics based on the established classics of vintage studio recording, including the Neumann U47 and U67 and the AKG C12. While their last major upgrade brought improvements to

indicated by the addition of an ’S’ to the product name. Hence the CM7 becomes the CM7 S, the CM12 becomes the CM12 S and so on. Wunder say that the upgraded circuitry in the new mics will enhance the clarity and sense of dimension in their sound. The Suprema Line versions of the Wunder Audio mics, which start at $1995, will be available from June. Any existing mic can also be

the power supplies, transformers and capsules, this time Wunder Audio are upgrading the passive electronics right across the board. Dubbed the Suprema Line, this new generation of mics will be

upgraded to Suprema status by contacting Wunder directly. Vintage King Audio +1 888 653 1184

Big score

Steinberg rewrite the rules for notation software with Dorico


teinberg have announced further details of their forthcoming brand-new notation software package, dubbed Dorico. Scheduled for release in the fourth quarter of 2016, Dorico is the product of a dedicated London-based R&D team, set up by Steinberg three years ago to develop what the company are calling the next generation of professional scoring software. Starting from the ground up, the developers have clearly put a lot of thought into workflow design, and Dorico is based around a single-window interface with a variety of page layout

and then impose a time signature and rebar the score at any point. Built-in grammar-checking algorithms allow you to freely insert music within an existing passage or adjust the durations of existing notes and rely on Dorico to automatically adjust the notation, although these auto-correct features can naturally be turned off for those who don’t want them. Engrave mode allows you to tweak and finesse the finer details of the score, while Print mode offers a full range of desktop publishing-style functions to achieve the desired layout and pagination. Play mode borrows Steinberg’s existing audio engine, with VST 3 plug-in support and a range of

options. Its five distinct operating modes mirror the usual stages of a project — Setup, Write, Engrave, Play and Print — and Dorico features what Steinberg claim is the most flexible note input and editing functionality of any scoring application. Interestingly, the software allows you to work entirely in open meter

VST instruments and effects immediately at your disposal, including HALion Sonic SE and the complete HALion Symphonic Orchestra Library. This impressive-looking package is set to cost $579, with educational discounts and crossgrades from Sibelius and Finale also available.

Ultimate weapon

McDSP release modular 6050 Ultimate Channel Strip plug-in


ith the 6020 Ultimate EQ and 6030 Ultimate Compressor, software experts McDSP took a rather different approach to these core processing elements, designing individual plug-ins within which a range of different processing modules, each with distinct sonic characteristics, can be freely arranged. Now they have combined the two to create the 6050 Ultimate Channel Strip plug-in. The 6050 contains all 10 EQ modules from the 6020 and all 10 compressor modules from the 6030, while adding eight new modules to the mix. These include upwards and downwards expanders, a dbx-inspired noise gate and a reissue of McDSP’s venerable FilterBank E4 module. Meanwhile, for those wishing to add a bit more colour to the proceedings, the new saturation and distortion modules should fit the bill. The user-friendly interface allows you to drag and drop modules to construct and reorder the signal chain of your virtual channel strip. The module selector panel allows you to quickly audition the individual modules, and any three modules can be placed in any order. Main input and output level controls with clear metering top and tail the three module slots, which have individual side-chain, solo and bypass controls. The 6050 Ultimate Channel Strip is available now in HD (AAX DSP/Native, AU and VST) and Native (AAX Native, AU and VST) bundles, priced $329 and $229 respectively. The new plug-in also forms part of McDSP’s Everything Pack v6.2. Owners of version 6 can upgrade for $199 for HD and $99 for Native. McDSP +1 650 963 9740

Tower of power

Barefoot Sound erect the MasterStack12 modular tower monitor


xtremely high-end and extremely highly rated, Barefoot Sound’s range of unique active monitors have successfully bridged the gap between big and small, delivering main monitor-style sound levels and performance from a nearfield monitor position. They now appear to have taken this idea to its logically conclusion by creating the MasterStack12, a 4.5-way active monitor tower that aims to deliver all the monitoring a professional studio will ever need in one imposing pillar of power. Previously only available privately to first-call Barefoot clients like Chris Lord-Alge and Kevin Augunas, the MasterStack12 is now open to all. All who can afford it, anyway, and at $47,525 per pair, we’re taking about professional facilities only. Still, the MasterStack12 is a fascinating prospect. Comprised of the MiniMain12 and MicroSub, sat on top of an input and power distribution unit with adjustable spikes to isolate it from the floor, the MasterStack12 stands 1.6 metres tall and weighs in at 127kg per tower. Each stack boasts a total of 10 individual drivers, each designed with an under-hung voice coil for extremely long linear excursion. The four side-firing subwoofers, three woofers and two mid-range drivers feature aluminium cones and are said to operate as perfect pistons to at least two-and-a-half octaves outside their respective bandwidths. A single ring-radiator tweeter delivers detailed high frequencies across a wide dispersion arc. Working in harmony, the precision drivers and sealed enclosures cover a frequency range of 18Hz to 50kHz with what Barefoot describe as “vanishingly low distortion, breathtaking dynamic range and ultra-fast transient response”. Four separate amplifiers, ranging from 250W for the tweeter to 2400W for the subs, supply the power. The ace up the MasterStack12’s sleeve, and the reason it can claim to be the last gigantic stack of incredibly high-end monitors you will ever need, is Barefoot’s DSP-based MEME (Multi Emphasis Monitor Emulation) technology, accessed via a convenient four-position rotary switch on a wired remote. Set to ‘Flat’, the system delivers the kind of revealing performance required for mixing and mastering. The ‘Hi-Fi’ setting is warmer and sweeter-sounding, emulating high-end domestic audiophile reproduction. ‘Old-School’ specifically mimics the sound of

a set of Yamaha NS10Ms, while ‘Cube’ emulates the mid-range emphasis of that other meterbridge staple, the Auratone 5C, allowing engineers to focus on potentially problematic areas of the mix and check real-world compatibility. All in all, it’s an awesome proposition. Barefoot Sound +1 503 894 8602

Nine lives

Propellerhead announce details of Reason 9


nitial details of the latest version of Propellerhead’s wildly popular music production suite Reason have been revealed. Reason 9 will feature a range of new devices, new tools and additional sounds designed to further extend the capabilities of this all-in-one music production platform. Highlights include pitch-editing capabilities for recorded audio and the ability to convert audio to MIDI. Reason 9 also introduces a trio of new Player devices, which perform real-time transformations on any MIDI input. Note Echo creates rhythmic, pitched MIDI delays and is designed for melodies, drum rolls and more. Scales & Chords will automatically generate harmonies and chords from a simple lead line, while Dual Arpeggio is a sort of super-charged arpeggiator, transforming chords into rhythmic runs, from classic up-and-down arpeggios to intricate polyphonic and polyrhythmic patterns. While Reason has featured audio tracking capabilities for some time now, the new Pitch Edit mode introduces a new level of sophistication. Particularly applicable to vocals, this

new set of tools allows you to fix tuning and timing and adjust dynamics but also get more creative, adding vibrato or creating a completely new melody. The new Audio to MIDI function, meanwhile, will let you transform your vocals still further, automatically turning the recorded audio into MIDI note data that can be sent anywhere you like. Reason 9 will also come equipped with 1000 new sounds, a range of workflow tweaks and a selection of new theme options. Available from June 21st, Reason 9 costs $449, with upgrades from any previous version available for $129.

Slave to the rhythm

Output unveil Movement rhythmic effects engine


aving shown what they can do with sample-based instruments like the inspiring and unconventional Exhale, Signal and Rev, Output have released their first effects processor. Movement is all about rhythmic effects and is designed to be applied to both individual sounds and instruments, and to entire tracks. Intended for either studio or live use, it features an array of different effects — including filters, EQ, delay, distortion, compression and reverb — all linked to a powerful rhythm engine that can be set to modulate any parameter in real time. There are in fact four separate rhythm engines inside Movement, each of which can use an LFO, step‑sequencer pattern, side-chain or the proprietary Flux and Randomizer as their modulation sources. The modulation destination is

Mac attack

entirely up to you, with a wide range of effects and parameters to choose from. Though Movement is clearly capable of some very complex polyrhythmic effects, the designers have made setting up patches as simple as possible, with an intuitive drag-and-drop mechanism for applying any one of the rhythm engines to any parameter knob on the GUI. To make things even more intuitive for live performance, an XY macro pad can control up to 152 parameters at once, while a global wet/dry mix control lets you fade the whole effect in and out. As you would expect, everything can also be fully automated within the DAW environment. With 300 presets on board to spark your imagination, it looks like another extremely powerful creative tool from Output. Movement is out now, priced $149.

Cakewalk set sights on Mac with Sonar OS X


fter many years of rumours, Cakewalk have announced an alpha testing phase for Sonar OS X, marking the first time the DAW has been available on the Mac platform. Clearly, the move to Mac is rather a momentous one for the makers of Sonar. The DAW has a long history of PC-only compatibility and deep Windows integration, but according to Cakewalk CTO Noel Borthwick, it was “increasing interest in Sonar from the Mac music creation community” that really got the ball rolling. Public testing is scheduled to start in Autumn 2016 and users can expect “100‑percent project file compatibility”. The alpha version will be

Featuring two different dynamic capsules in one mic, the DX-2 from MXL is an intriguing new device designed for guitar cabs, horns, drums and other instrument

free to use and will include a subset of Cakewalk plug-ins and modules for the ProChannel, Sonar’s acclaimed, per-track modular channel strip that resembles an analogue console and includes emulation of three vintage consoles. Cakewalk are hoping Sonar will stand out in the crowded DAW space on Mac with features like an SSL-style compressor on every bus; Mix Recall, which takes snapshots of a mix ‘scene’ and saves the snapshots within the project; a flexible interface letting you show and hide menus with shortcuts, dock many items, and move modules around to your liking; and deep Melodyne integration. The Mac version supports the same plug-in standards that Sonar Windows supports: VST2, VST3 and DX. Cakewalk plug-ins in these formats are supplied with the Alpha version, so users will be able to experiment with plug-ins in Sonar OS X. Cakewalk have also announced that any purchase or upgrade of Sonar Platinum until August 31 will now include lifetime updates. These updates will include planned feature enhancements including ripple editing, plug-in load balancing for improved CPU performance, and new comping and take-management features.

It’s been a long, long time since we’ve been able to bring you news of new developments in the world of magnetic tape! Enter French company Mulann, who have revamped their pro audio tape range, now rebranded as Recording The Masters. Mulann owns the original formulae for magnetic tape created by Agfa and BASF, and uses original BASF/EMTEC

compact. Featuring a robust metal body, the LS-P2 is less than 15cm tall and just 1.8cm thick, ideal for slipping into a jacket pocket or mounting on top of a digital camera. Recording in space-saving MP3 format or in WAV format at up to 24-bit/96kHz, it records to 8GB of internal flash storage, expandable using MicroSD cards, and

sources. Featuring a blend knob on the body of the mic itself, the DX-2 lets you select one capsule or the other, or mix between the two. Capsule 1 is a large, supercardioid design that will pick up more room ambience, while Capsule 2 is smaller with a cardioid pickup pattern designed to reject sources to the sides and rear. Beyond the difference in pickup pattern, there should be some tonal differences to play with, too, and MXL say that the two capsules can be successfully blended without significant phase cancellation. The new mic will be on display at the Summer NAMM show and is set to be released on August 1st, with pricing yet to be announced when we went to press.

plant equipment. The Recording The Masters range includes a variety of tape formulae designed for professional recording and mastering, as well as semi-pro, audiophile and archiving applications. Officially launched at the AES Convention in June, Mulann say that the RTM tape will offer very high sensitivity coupled with a long working lifespan. http://get.recording

Aimed at sound recordists and videographers, the new LS-P2 recorder from Olympus is extremely

features a built-in USB connector that extends out of the casing to directly connect to a computer. Unusually, the LS-P2 features a trio of integral mics, with a third situated between the main stereo pair, which is said to extend the recorder’s frequency range down to 20Hz. The LS-P2 can also connect directly to Bluetooth peripherals such as speakers and headphones, while the included rechargeable battery is said to give it up to 39 hours of recording time. The LS-P2 is available now priced $179.99.


Off The Record Music & Recording Industry News

How can we expect people to value music when we don’t keep track of who owns it? DAN DALEY


he death of Prince in April stunned everyone. But it also revealed a huge elephant in the room. The Artist Formerly And Then Known Once Again As Prince left behind a massive trove of music in varying states of completeness. Worse, it is almost certainly also in various conditions of ownership. Somewhere among the 26 LPs’ worth of material, he reportedly left behind a slew of documentation — or not — of when and where and under what conditions certain recordings took place. That will have a huge effect on the value of those recordings, and of the estate itself. Early estimates put it at around $300 million, but the ‘celebrity death boost’ — the bump in sales that accompanies the passing of a notable — could push it as high as $800 million. Nielsen reported that on the day he died, grieving fans bought 239,000 Prince albums, with another 399,000 albums sold that weekend — up more than 16,000 percent from the previous weekend, when about 2500 albums were sold.

The Reel Deal How Prince’s career will proceed from here will be an interesting story, especially if you’re a lawyer. It will likely follow the arc of Elvis Presley’s, and those of Jim Morrison, Jimi Hendrix, Marilyn Monroe, James Dean and a host of other celebs who have

earned far more money dead than alive. But Prince’s recorded treasure trove at Paisley Park underscores a related phenomenon that’s plagued studios for going on a half a century: production elements left behind at the studio, sitting in legal and creative limbo while collecting both dust and the potential for some very annoying liabilities. For much of that time, studios would end up storing dozens or hundreds of multitrack reels — basic tracks, vocal‑comp reels, drum‑track reels — and quarter‑ and half‑inch mix reels, left behind as the detritus of production but still retaining huge potential value, depending upon the artist. More recently, hard drives, thumb drives, FireWire drives and other formats lie in boxes on shelves, awaiting reclamation by their creators. Each of them is a possible treasure, but one fraught with unknowns, both creative and legal. As attorney Elizabeth Gregory, who has represented dozens of artists and producers, told me: “The big problem is that often there are incomplete or even no production notes at all left behind. You don’t know who played on the masters, whether the players have been paid, whether the masters contain samples, who wrote what and what the splits are, who produced, who mixed them, and so on. Then there are side‑artist clearance issues, union issues, copyright issues, possibly exclusivity waiver issues regarding guest artists. You may not know when they

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were recorded. Maybe they were created during the term of an exclusive recording or publishing agreement, and subject to the terms of those agreements. It’s terrifying. At least with digital recordings you can usually determine when they were created, but as to who is on them? Good luck.”

Who’s Got What? Orphaned production elements have been a problem since time immemorial (only 2 percent of the 3000‑plus cylinder recordings produced between 1889 and 1894 by the North American Phonograph Company have been accounted for). In the age of tape, bulky boxes of Ampex and BASF multitracks used to clog studio closets and storage rooms until they reached some kind of critical mass, prompting studio managers or owners to threaten forgetful clients with bulk erasure if they weren’t retrieved. These remnants of decades’ worth of sessions have become the stuff of legend. Bill Holland, Billboard’s Washington bureau chief in the 1980s and ‘90s, recalls on his blog a moment in the ‘70s when Columbia Records decided to clean out their storerooms. “So the word gets out, and at lunchtime, there’s all sorts of people from the building standing around the dumpsters on East 52nd St, and there are tapes, acetates, track sheets, session notes, you name it, all over the street, and people trying to save this or save that,” someone present at the time remembered. The two largest labels of the end of the tape era, Sony and BMG, were reported to have 1.9 million tapes, acetates, metal parts, and so on between them. Small wonder that independent studios had an even harder time of it, especially in New York City, where space is always at a premium. Dave ‘Roz’ Rozner, who has managed

Quad Recording there through its various iterations over three decades, says clients, whether they’re major labels or indie artists or producers, have not gotten any better about managing their materials. “We’d call them, fax them, message them, take out an ad in the newspapers, then we’d have to purge,” he says. “It used to be tapes; now it’s tapes, hard drives, thumb drives, everything. Some clients come in with a two‑track and just overdub vocals, then leave with a mix on a thumb drive and leave the rest behind. It’s become so disposable.” Until it’s not. Tony Drootin, who managed MIDI pioneer facility Unique Recording for 13 years, Sony Music Studios for a decade, Puffy’s Daddy’s House Studio for five years and now runs Platinum Studios, says there are gems in those piles of production elements but they’re buried under the weight of time, lack of documentation and simple neglect. “It only got worse as productions became more complicated,” he says, using Michael Jackson’s Invincible as an example. “They might have recorded 35 songs but only 12 get picked for the album. The rest get scattered, track sheets disappear. No one remembers who produced this or who wrote that.” As artists leave us, their posterity often takes on added value. Those scattered production elements are no longer just detritus but potential gold nuggets of the future. The bottom line is that leaving pieces of projects scattered about is both leaving money on the table and asking for trouble later. Worse, it further encourages the forces that conspire to devalue music, in an age when it’s too easy to both make and consume. Pick up your stuff. It sounds like your mother talking, but she did turn out to be right, didn’t she?

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Pioneer RM‑07 Nearfield Active Monitors

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Pioneer might not be the first name you’d associate with studio monitoring — but their new creation is quite remarkable. PHIL WARD


f you’d asked me, prior to the announcement last year of the RM‑07, if I was ever likely to express anticipation about a Pioneer active monitor, the answer I think would have involved a raising of the eyebrows. However, since the launch of the RM‑07 (and the smaller RM‑05), I’ve been genuinely looking forward to getting my hands on a pair. So, what’s going on? Well, first, what do we generally know about Pioneer? If, like me, you grew up a spotty hi‑fi geek in the ’70s, you’ll certainly remember the PL12D turntable — almost invariably partnered with a Shure M75 cartridge, and the almost obligatory upgrade from the beginners’ Garrard SP25 MkIV. And

Pioneer RM‑07 $1598 PROS

• Consistent dispersion and imaging from a quality compound driver. • Natural, un‑hyped tonal balance. • All the qualities needed for a trustworthy mix tool. CONS


Proper attention to electro‑acoustic detail delivers the goods. The RM‑07 is a classy and well‑engineered monitor that delivers genuine quality and engineering integrity at an entry-level price.

then, in the early ’90s, among Pioneer’s huge range of hi‑fi separates, the A400 was for a while widely considered the best‑sounding mid‑level integrated hi‑fi amplifier. Moving up to date, Pioneer of course have a profile in the DJ sector, with their range of CD decks, mixers and headphones and, actually, a couple of existing but somewhat underwhelminglooking active nearfield monitors. But the company also have a secret: TAD Labs, short for Technical Audio Devices Laboratories (http://www. TAD Labs are a US‑based offshoot company, a sort of Pioneer ‘skunkworks’ organisation, that design and manufacture a range of very high‑performance hi‑fi speakers, source components, amplifiers and PA drivers. And according to the marketing information, the RM‑07 had significant TAD Labs design input. TAD Labs’ existence is not the only secret, however. Also not widely known is that, until very recently (and almost certainly during the RM‑07’s development), the Director of Speaker Engineering at TAD Labs was a Welshman called Andrew Jones. Jones is steeped in British electroacoustic design traditions and values. Following a Physics degree from Surrey University, he joined KEF Electronics and learned his electroacoustics and speaker engineering skills under speakerdesign doyen Laurie Fincham (for a touch of background on Fincham, there’s a good interview here: index.php/interviews‑hi‑fis‑celebrities/ laurie‑fincham‑thx‑2007/). When the

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original KEF company foundered in the mid ’90s (they were bought, and are still owned, by Hong Kong‑based Gold Peak Industries) both Fincham and Jones moved to the US and joined Infinity (www. Their paths only parted when Jones moved on to TAD Labs. But what, I’m sure you’re asking, has this got to do with the Pioneer RM‑07? Well, during Jones’ time at KEF the company developed and launched their groundbreaking and patented ‘Uni‑Q’ dual‑concentric driver technology — and not coincidentally, I suspect, the Pioneer RM‑07 features a similarly configured compound driver.

slightly dumpy of proportion, with a bulging of the sides and softening of the edges. The proportions don’t quite work aesthetically to my eyes and they’re not really helped by the unremittingly matte‑black finish. I think something grey and classy might have worked better. And don’t get me started on that gauche, italicised Pioneer logo. I know it’s been that way for decades, but I’ve never warmed to it. Despite my aesthetic misgivings, the enclosure shape is an acoustically benign one, and its construction, fully in die‑cast aluminium, gives it a rigidity and non‑resonant nature that flat-panel,

What A Coincidence Describing the KEF Uni‑Q technology as ‘dual‑concentric’ is actually not entirely correct; ‘dual‑coincident’ is more accurate. The difference may appear at first to be one of semantics, but it actually describes a significant variance between the KEF technology that looks, to my eyes, to have inspired the RM‑07 driver, and the dual‑concentric driver technology that went before it; the best known of course being from Tannoy. ‘Dual‑concentric’ implies that two ‘elements’ are located on a common centre line, whereas ‘dual‑coincident’ implies that the two elements are, effectively, in the same location in space. I’ve explained this, and why it is significant, in the ‘Short History Of Dual‑concentric Drivers’ box. The single, compound driver of the RM‑07 results in a speaker that looks a little different from the usual fare of inexpensive two‑way active nearfield monitors. The RM‑07’s enclosure is July 2016 / w w w . s o u n d o n s o u n d . c o m


wood-based enclosures can only dream of. It is truly built like a brick cliché. The aluminium enclosure also endows the RM‑07 with some significant weight. If there were a league table of kilograms per unit price for nearfield monitors, I’ve little doubt the RM‑07 would be near the top. They felt an awful lot of speaker for the money when I lifted them on to my hyper‑sturdy wall brackets. The compound driver comprises a nominally 150mm-diameter woven Aramid fibre cone bass/mid element and a 30mm aluminium dome HF element. An upturned skirt on the HF dome is said to increase its rigidity so that the driver’s frequency response extends to 50kHz. The active crossover between the two drivers is at an unusually low 1.6kHz, which leaves me wondering a little whether the HF driver might not be somewhat prone to thermal compression as its voice coil warms up (because its voice coil will undoubtedly warm up more than it would if the crossover was more conventionally an octave higher). Amplification is provided by a rear‑mounted module specified at 100 Watts for the bass/mid driver and 50 Watts for the HF driver. The amplification technology is traditional Class‑A/B and its power supply is a conventional linear design with a toroidal mains transformer. Along with an IEC mains socket, rear‑panel connectivity comprises just analogue balanced XLR and unbalanced phono input sockets. A gain control and some LF, MF and HF EQ options are also provided. I’ll describe the EQ options in more detail a little later, but I’ll have my usual moan about variable gain controls without detents now: you

see, unless the gain is at maximum or minimum, it’s difficult to be sure that both speakers of a pair are gain-matched. The last remaining rear-panel feature is an auto‑standby switch. When ‘on’, the RM‑07 switches to standby mode after it’s been silent for a while, and switches back on automatically when it detects an input signal. I used the feature for a while but soon switched it off — I found it sometimes frustratingly slow to switch on.

Pass The Port Back around the front of the RM‑07, beneath the driver is a wide, letterbox‑shaped reflex port that incorporates a few interesting and very

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unusual refinements. Regular readers will know that I am, by default, suspicious of reflex ports, preferring the simplicity and more accurate time‑domain performance of closed‑box speakers. If there has to be a reflex port, however, there are good and bad ways of implementing such things, and the RM‑07 port falls firmly in the former category. Firstly, it appears to be tuned relatively low — around 45Hz. This means its real-world impact on time‑domain performance is likely to be more benign that would be the case if it were tuned, as some are, significantly higher. Secondly, there are signs in the design of the port that effort has been expended on ensuring airflow remains linear, and in particular not coloured by a mid‑range ‘organ‑pipe’

resonance (see the ‘Pipe Down’ box). A particularly fascinating little feature that Pioneer call AFAST (Acoustic Filter Assisted System Tuning), revealed in the cutaway drawing of the RM‑07 above, is designed to suppress the fundamental organ-pipe resonance. AFAST consists of a short, closed ‘pipe’ that feeds off the port tube. My guess is that the AFAST pipe is positioned along the port tube at an anti‑node location of the potential organ‑pipe resonance to provide a counter‑resonance. Think of it as the acoustic equivalent of a mass damper. Fascinatingly, a technology piece I read a little while ago on the 2014 Ferrari F1 engine described its use of a very similar technique to modify the resonant

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behaviour of its exhaust pipes. I wonder who came up with the idea first or if their similarity isn’t coincidental? AFAST isn’t the only refinement of the RM‑07 reflex port. At the port mouth there’s some complex flare profiling intended to manage airflow dynamics in order to delay the onset of noisy and non‑linear turbulence, and within the port, it’s just possible to see a few adhesive damping pads stuck to the inside surface. These presumably are present to suppress mechanical resonance or, possibly, to further modify airflow dynamics. These various port refinements are genuinely unusual and point to the design of the RM‑07 being by somebody who lives and breathes electroacoustics. Andrew Jones is, of course, prime suspect, and we enquired of Pioneer but they were unable, or maybe unwilling, to confirm Jones’ identity as designer — perhaps because he recently departed TAD Labs to join ELAC of America.

on that front because, cutting directly to the chase, the RM‑07 is good. In fact it’s really, really good — way better, indeed, than you’d expect at the price. To begin, with I was a little underwhelmed, but the more I listened, and especially as I listened to higher‑quality material, rather than the Spotify stream that was playing when I first set up the RM‑07s, the more I began to appreciate what I’d describe as its understated accuracy. With its EQ

Listening In Having a monitor perhaps designed by an engineer who lives and breathes electroacoustics doesn’t necessarily guarantee it’s going to sound any good — but we can relax July 2016 / w w w . s o u n d o n s o u n d . c o m


options flat, the RM‑07 has a slightly warm, almost BBC‑like tonal balance, so it doesn’t immediately seduce with overt clarity and detail. But as I tuned in to its qualities I began to appreciate that all the clarity and detail is present and correct, it’s just presented in a neutral and undemonstrative manner. The RM‑07 is not one of those nearfield monitors that will prove tiring with its insistent “listen to me!” character. Rather, it will just let you get on with your job, while it does its job of presenting reliable, consistent and informative audio. All the way from bass through mid to top, the RM‑07 offers a coherence of character that sounds detailed and trustworthy. Along with its generally coherent and ‘together’ character, the RM‑07 provides all the qualities you’d hope to hear from a well-sorted compound driver. Its dispersion, both vertically and horizontally, is wide and consistent, and its portrayal of stereo imagery both laterally and in terms of depth is focussed and stable. The RM‑07’s slightly warm default balance is perfectly usable, but if your preference is to modify it a little, the rear‑panel EQ options are well chosen to provide the right kind of subtle balance tweaks. The mid‑range cut control, for example, is unusual but it works well. Not only is its operational frequency rather lower than usual (140Hz — right in that male voice band, just above bass, where rooms tend to misbehave), but as more mid is cut, the Q of the notch filter increases. The LF and HF options provide more conventional characteristics, but are again well chosen in terms of levels and frequencies: the shelf LF filter provides potentially useful compensation

Alternatives It’s not front‑page news that the world isn’t short of active nearfield monitors at or around the same price as the RM‑07. Dynaudio, Adam, Genelec, PreSonus, Focal, Yamaha, Event, Eve and numerous other manufacturers can all offer competition for the Pioneer RM‑07. Take your pick.

for boundary locations (especially when combined with the MF filter), and the HF shelf filter offers a little more or less ‘air’ at the very top end. And speaking of the top end, the HF driver displays the same slightly undemonstrative nature as the rest of the speaker, but it’s still usefully detailed and uncoloured. At the bass end of the bandwidth, while the RM‑07 is unmistakably a ported monitor, I had no concerns that I was listening to an over‑hyped or poorly sorted reflex alignment that trades LF bandwidth against value as a reliable mix tool. It seems to me also that a designer who has gone to so much effort to ensure the reflex port is linear is unlikely to be greedy of LF bandwidth only to end up paying the price in the time domain.

Lend Me Your Pioneers I’ve been lucky enough to review in these pages a few extravagantly priced monitors with sky‑high aspirations. The Pioneer RM‑07 is rather different, however, firstly in that it’s priced at around the entry level for serious active monitors, and secondly because it doesn’t really make claims to break revolutionary technological ground or overtly aspire to be the best. But of all the recent nearfield monitors I’ve tried it’s actually my favourite — not just because it works so well and fundamentally

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does the job, but because it seems to me to represent genuine, thoughtful, ‘old‑school’ and skilful electroacoustic engineering of great integrity and quality. If I wanted to recommend it more highly, I wouldn’t know how. And no, I really

never imagined I’d be writing that about a Pioneer speaker. $$ $1598 per pair. TT Pioneer Electronics USA +1 800 782 7210 WW

A Short History Of Dual‑concentric Drivers Perhaps the first question to answer in any discussion of dual‑concentric and dual‑coincident ‘compound’ drivers is: what’s the big deal? Well, the big deal is that in any speaker system in which the audio band is covered by multiple drivers, the problem of ‘crossing over’ from one driver to the next is exacerbated if they’re not in the same position in space. The path length, and therefore acoustic ‘flight time’, from each driver to the ears changes as the listener moves, and due to the resulting phase changes, the system frequency response also wanders about. This unavoidably applies to all speakers with displaced multiple drivers, even ones advertised as ‘time aligned’ or ‘linear phase’. They will only fit that description at a discrete set of positions in space (the corollary of which is that speakers not advertised as time aligned or linear phase are actually likely to be just that at a different set of discrete positions in space). Conventional displaced multiple‑driver systems also tend to to be characterised by an awkward dispersion discontinuity, where the narrowing radiation of the bass/ mid driver hands over to the initially much wider radiation of the HF driver. So, arranging for multiple drivers to be dual‑concentric or, even better, dual‑coincident, potentially wipes away

two of the fundamental technical issues of multi‑driver systems. The classic Tannoy dual‑concentric driver (actually five years pre‑dated by Altec Lansing) is arranged as a conventional bass/mid driver with a high‑frequency driver mounted on the back of the magnet and radiating through a hole in the pole‑piece. Diagram 1 illustrates a simplified version of this topology. While the Tannoy solution achieves the basic aim of drivers on the same centre, albeit with the HF driver significantly behind the bass/mid driver, it has a fundamental weakness in that an HF driver radiating through the pole piece can only really be horn loaded — in effect, it’s a PA‑style compression driver. Horn loading of course brings increased radiation efficiency, but it also results in narrow dispersion and often higher levels of coloration (particularly if the horn shape is compromised by also playing the role of pole‑piece and bass/ mid diaphragm). Laurie Fincham’s KEF dual‑coincident solution was driven by the development of rare‑earth permanent magnets in the ’70s. Fincham realised (before anybody else, hence the patent) that, with a neodymium‑iron‑boron magnet, a high‑frequency driver could be made both small enough and sensitive enough to sit on the end of a bass/mid driver

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pole‑piece, where it would be both on the centre‑line and at, or at least very close to, the acoustic centre of the driver. Diagram 2 illustrates this topology. I wrote that these compound driver topologies “wipe away two of the fundamental technical issues of multi‑driver systems”, and they do. But of course, as is the way with speakers, they introduce some issues of their own. The first is that, even with the KEF arrangement, the dispersion characteristics of the HF driver are unavoidably influenced by the bass/ mid diaphragm. Unless the the bass/ mid diaphragm is made flat there is a degree of ‘horn’ loading inevitable from its conical surface (Technics do actually manufacture a flat‑diaphragm dual‑coincident hi‑fi speaker). This means there’s a compromise to be managed between the diaphragm shape required to achieve the desired bass/mid driver characteristics, and that needed

to optimise the performance of the high‑frequency driver. The KEF approach has been to develop the dual‑coincident high‑frequency dome and bass/mid driver so that they work together to create a seamless compound driver with close to ideal dispersion characteristics; and while their original basic patent has now lapsed, some more recent patents protect significant aspects of this technology. A second potential issue that can arise with compound drivers, at least in theory, is high‑frequency intermodulation distortion caused by movement of the surrounding and adjacent bass/mid diaphragm modulating the output of the high‑frequency driver. Imagine for a moment that the bass/mid cone is moving backwards and forwards at, say 80Hz, and the high‑frequency driver is simultaneously playing a 3kHz tone. Modulation of the 3kHz signal by the 80Hz signal will potentially

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result in signals at 3.080kHz and 2.920kHz (the sum and difference of the two frequencies). The level of the intermodulation distortion increases with movement of the bass/mid diaphragm, but its audibility is not cut and dried. Some claim it’s a deal‑breaker, although those that do tend not to have the technology or the means to develop it. Personally, I reckon such distortion is potentially audible when a compound driver is used in a small two-way system. My own pair of KEF LS50s, for example, can, I think, sound slightly cleaner at the top end when their reflex ports are left open so that low‑frequency bass/mid diaphragm movement is reduced.

One technique, and the one used on the RM‑07, for potentially reducing both intermodulation distortion and generally the influence of the bass/mid diaphragm on a concentrically mounted HF driver, is to constrain the dispersion of the HF driver a little with its own waveguide. While there are potential down sides to this technique — the dual‑coincident nature may be degraded, and dispersion discontinuities between the two drivers are likely to become more pronounced — it perhaps has a further advantage for Pioneer in that it partly negates the risk of infringing any of the more recent KEF patents covering the close integration of coincident drivers.

Pipe Down! When we talk of reflex port resonance we think primarily of the of the ‘slug’ of air in the port bouncing against the spring provided by the air in the enclosure. This is a Helmholtz resonance (after the German physicist and philosopher) and is the one that a ported speaker employs to extend

low‑frequency bandwidth. But many ports demonstrate a second, much less desirable resonance higher up in frequency that’s analogous to that of an organ pipe (or any other wind instrument really). Typically, and depending on the length and diameter of the port tube, the organ‑pipe

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resonance can be anywhere between, say, 300Hz and 1.5kHz. It will usually be a relative high‑Q resonance and can reach a level as loud as the desired output of the port. Needless to say, port tube organ‑pipe resonance is not a good thing! To illustrate this phenomenon, I generated a couple of frequency response curves using FuzzMeasure Pro. The curves were generated by placing a measurement microphone right at the mouth of the port and, and as such, will be contaminated to some extent by the output from the adjacent bass/ mid driver; however, in my experience

such measurements can provide reliable comparative data. Graph 1 shows the port output of the RM‑07, from 20Hz to 2kHz. The Helmholtz resonance is obviously revealed at around 45Hz, above which the output decays into ‘noise’ with only a couple of mild potentially resonant features apparent. Graph 2 shows a similar measurement for an unnamed monitor with a similarly tuned and similarly proportioned port. This port has no features obviously designed to suppress organ‑pipe resonance, and displays significant resonant features at around 480Hz, 700Hz and a little over 1kHz.

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Legendary Keyboards Reinvented


REINVENT 17 authentic recreations of the legendary synths, organs, electric pianos and more that made music history—enhanced to help you make tomorrow’s music.


Tom Oberheim

Two Voice Pro Analogue Synthesizer

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Tom Oberheim hasn’t just revisited his classic Two Voice synthesizer, he’s made it even better. GORDON REID


berheim’s SEMs (Synthesizer Expansion Modules) were first manufactured in 1974 to allow users to play sequences using a small and relatively affordable sound generator rather than tying up a larger and more expensive synth such as a Minimoog or ARP 2600. People really liked the sound of the little box so, when Oberheim began building various numbers of them into cases that featured one or more keyboards and other paraphernalia such as programmers and sequencers, the larger models became some of the most revered synths of the era. But the smallest model, the 2‑Voice, was a bit of a mythical synth — not because it was over‑endowed and had an unhealthy interest in virgins, but because few players ever laid eyes upon one. You could occasionally gawp at grainy monochrome photos of the unwieldy 4‑Voice and the gargantuan 8‑Voices in the music rags of the era, but the little 2‑Voice remained under the radar. This was a shame because its dual SEMs made it an unusually flexible synth capable of generating sounds that you couldn’t obtain from any other integrated synth of the era. You could also use its ‘mini sequencer’ to control the SEMs, sequencing SEM A while playing SEM B, or vice versa, or sequencing both simultaneously. All of this was packaged in a case with its own lid for transportation, and I have long wondered why the 2‑Voice wasn’t adopted more widely by the likes of Tangerine Dream (who used

many other Oberheim synths) or Neu!, or anyone else pushing the boundaries of electronic music at the time. Whatever the reason, it had disappeared by the end of the 1970s and, despite resurfacing occasionally in the hands of (for example) Liam Howlett of the Prodigy, it remained largely unknown for the next 30 years. But that changed when, in March 2012,

Tom Oberheim Two Voice Pro $3495 PROS

• It’s convenient, simple to use and sounds superb — an ideal combination. • It’s semi‑modular, and offers extensive connectivity. • Unlike the original 2‑Voice, it offers velocity and aftertouch sensitivity plus pitch-bend and modulation wheels. • The sequencer is considerably enhanced when compared with the original. CONS

• It’s not as fully featured as some of the alternatives. • It lacks patch memories. • It would benefit from a more comprehensive manual. • It’s not cheap. SUMMARY

The TVS Pro has its limitations and, given its price, won’t appeal to everyone. But it’s stylish and capable, has a distinctive character, and can sound superb. It’s probably destined to become a modern classic.

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Tom Oberheim announced his intention to release an enhanced version of the 2‑Voice. As before, the TVS Pro would be based on two SEMs married to a mini sequencer and a three‑octave keyboard, but would feature numerous significant enhancements that would drag it into the 21st Century. And four years later (!) here it is...

The SEM Sound Generators The SEM is a relatively basic synth and, if you understand how the audio and control signals are routed with it, it’s hard not to obtain interesting or useful sounds from one. But this doesn’t mean that it’s all‑encompassing — far from it, in fact. SEMs may have been responsible for some of the warmest sounds in synthesizer history, but you’ll not get one to imitate the Minimoogs or ARP2600s that it was designed to liberate, nor even the simpler Japanese synths of the era. Each of its two oscillators generates just two waveforms (sawtooth and pulse/ PWM waves) and you can only access one of these at a time. Low-frequency modulation is even simpler — just a sine wave. Similarly, the SEM retains ADSD

contours, where the Decay and Release times are always the same. But if it has a trick up its sleeve (and it does) it’s its 12dB/octave multi‑mode filter, which allows you to select a band‑pass mode or sweep from low‑pass through band‑reject (notch) to high‑pass responses. It’s hard to explain in words how different an SEM can sound when, for example, the filter mode is sitting halfway between low‑pass and notch, but it generates a sound that most other synths can’t, and that sound can be superb. Tom must have been tempted to update the SEM when he recommenced manufacture in 2010 but, when I reviewed the first of the stand‑alone versions (Sound On Sound, September 2010) he told me that he had designed it to be as close to the original as possible. At the time, I tested this by removing one of the original SEMs from my 4‑Voice and comparing the two. Although the boards looked different and there were minor differences in the controls, it was clear that the underlying circuitry was the same. To be fair, testing revealed that the minimum and maximum cutoff frequencies of the new unit’s filter were

Son Of 4‑Voice Shortly after re‑launching the stand‑alone SEM, Tom Oberheim announced that he would be releasing the SO4V ‘Son Of 4‑Voice’ synthesizer. With a projected delivery date in 2012, this caused huge excitement among the analogue synth community. But the original launch was missed, as was the next, and the next, until people started to speculate whether it would ever

appear. Then, at NAMM 2015, Tom told me that he had indefinitely shelved his plans for the SO4V, and that remains its status to this day. With the launch of the TVS Pro, you might be wondering whether you could add a couple of stand‑alone SEMs to this to create your own 4‑Voice, but I’m afraid that it isn’t possible; the keyboard decoder only generates CVs and Gates for two notes.

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different, as was its response to Env 2. There were also differences in the slowest contour rates, the maximum modulation depths and the maximum LFO rate, all of which meant that there were some sounds that could be programmed on the modern SEM and not the vintage one, and vice‑versa. But when I used the new SEM as part of the 4‑Voice it proved to be a perfectly acceptable substitute, and it would have been a braver man than I who claimed that he could always tell which was which. I’m therefore surprised when I read comments from elsewhere stating that new SEMs sound (and I quote) “nothing like the real thing” and that vintage modules sound “way better” than the modern ones. In fact, I think that they’re daft.

Connectivity An original SEM offered a total of just eight audio input and output, CV and Gate sockets. Sure, Oberheim provided instructions on how other functions such as filter CVs could be tapped from the main board but, in 2010, when the CV & Gate version of the modern SEM appeared, it provided no fewer than 33

patch points on its top panel. Today, the TVS Pro offers considerably more, and you might think that its 56 3.5mm top‑panel sockets and six quarter‑inch rear‑panel sockets include all of those from each SEM, minus those that would otherwise be duplicated. They don’t. For example, the second pitch CV input has been removed from each oscillator, as have the Sync In and Sync Out sockets. Likewise, the second CV input to the filter has gone AWOL, and the four filter outputs — low‑pass, band‑pass, high‑pass and notch — have been replaced by two, one for the multi‑mode low/notch/high signal, and one for the band‑pass signal. Furthermore, the Env 1 and Env 2 trigger inputs have been discarded, as has the VCA output, and the dual audio inputs per SEM have been reduced to a single input per SEM. There are, therefore, 19 patch points on the top panel per SEM, and the other 18 comprise two 1‑in/3‑out multiples plus 10 sockets carrying CVs and Gates generated by the keyboard and sequencer: CV A and CV B out; Gate A and Gate B out; plus mod wheel, S&H, vibrato LFO, velocity A and B, and aftertouch out.

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Around the back of the TVS Pro you’ll also find the standard complement of MIDI In/Out/Thru on five‑pin DIN connectors. Surprisingly, the synth only understands MIDI channels 1 to 7 (or 1 to 8 if you use the keyboard Split and Unison modes), and it both generates and responds to a very limited set of commands: Note On/ Off (with velocity), transpose (output only), pitch bend, modulation and aftertouch. Nonetheless, this is sufficient for many purposes and, as a bonus, the CVs and Gates generated within the interface are output from the appropriate sockets, so you can also use the TVS Pro as a simple MIDI/CV converter.

The Mini Sequencer The mini sequencer in the original 2‑Voice combined a simple two‑row, eight‑step (maximum) sequencer with a Sample & Hold generator. You could set up Seq A (hardwired to SEM A) and Seq B (hardwired to SEM B) and then use the assignment switches to select whether each of the SEMs was controlled by the keyboard, the S&H, or the relevant sequence. It wasn’t the most powerful sequencer available, not was it the most intuitive, particularly with respect to the ways in which sequences were transposed, or in the way that the sequence tempo could be controlled from the keyboard, so perhaps that was why Oberheim called it a ‘mini sequencer’. As you would expect, the sequencer in the TVS Pro has been considerably enhanced. It retains the two rows hardwired to their respective SEMs, but now with a maximum of 16 steps and, unlike the vintage synth, its output voltages are now quantised to semitones

rather than being continuously variable, so users will be able to set up musical sequences quicker and more easily than before. (It will also piss off the microtonal music crowd, who will have to look elsewhere.) Additional facilities include the ability to transpose sequences, to apply two‑, three‑ and four‑way ratcheting (multiple triggers on a given step), to insert rests, and to determine the gate length for all of the notes within a given sequence. Oh yes, and as one of a handful of Shift functions, you can set things up to start sequencing when you press a key on the keyboard. As well as generating CVs and Gates, the mini sequencer also talks MIDI, responding to MIDI Start and Stop commands as well as synchronising to MIDI Clock, although with a narrow range of clock divisions: one, half and quarter. It also sends MIDI Clock, Start and Stop messages as well as note information when it’s playing, the last of which is not always the case with integrated sequencers from elsewhere, and therefore welcome. However, the biggest difference between this mini sequencer and the original is the inclusion of a small flash memory that can hold up to 100 (50 sets of two) sequences. These can be used individually or chained into songs, with transposition and repeats of patterns, as desired. I found composition to be a bit fiddly and, initially, made many mistakes when trying to program songs, but it eventually seemed to work as intended. I’m confident that, had I had longer to work with this aspect of the TVS Pro, it would have become second nature. Just be prepared to swear a few times before you get to that point.

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In Use The first thing you’ll notice about the TVS Pro is how small and light it is. Unfortunately, unlike the original 2‑Voice, it’s no longer built into its own road case, but at least it still comes with an integrated power supply which, in these days of wall warts, is a relief. The key to using it starts with understanding the Assignments box in the mini‑sequencer panel, which determines whether either or both of the SEMs are to be controlled from the keyboard, by the sequencer, or by the S&H generator (see box) and, when the keyboard is selected, how its various unison, split and polyphonic sub‑modes work. The keyboard always responds with ‘last‑note priority’, so these determine how retriggering is affected (or not) by

legato playing, as well as whether the SEMs cycle on consecutive notes or whether SEM A can be held as a first note with all subsequent notes played on SEM B. Be aware, therefore, that the TVS Pro lacks the high‑note and low‑note priority options found on the standalone SEM and many other modern synths, which could be important if you’ve developed a soloing technique based upon one or another. Oberheim’s documentation makes a big thing about the SEM being analogue and ‘not just its signal path’, meaning that the contour generators and the LFO in each module are also analogue rather than being digitally generated and then converted to CVs before being applied to the VCO, VCF and VCA. From a sonic perspective, this doesn’t bother me one way or the other, nor should it

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you, although it’s obviously important with respect to the authenticity of the SEMs. Much more significant is the fact that secondary functions that could have slipped into menus are still controlled using knobs: the level of each SEM and its pan, the portamento time for each SEM, and the levels of the CVs generated by velocity and aftertouch. Strangely, the aftertouch knob doesn’t control the amount sent via MIDI, although the velocity knob does. I wonder why? Ah yes, the keyboard. Although your primary use for the TVS Pro may be as a sequence generator, for me the thing that raises it way above its ancestor is the provision of velocity sensitivity (with eight response curves) and aftertouch. The first of these is directed internally to the amount of Env 1 and Env 2 applied to the VCF cutoff frequency and the VCA Gain (which is the correct way of doing such things), while the second controls the amount of a dedicated vibrato oscillator applied to the oscillators’ pitches. But that’s far from the whole story, because they are also presented as a pair of CVs accessible from the top panel, allowing you to direct them to all manner of addition destinations. This is a huge step forward, and makes the TVS

Pro an expressive instrument in exactly the way that Oberheims in the mid‑1970s were not. Just as important in this respect is the panel to the left of the keyboard, which Oberheim calls the Bendbox. This contains the pitch‑bend (1 to 12 semitones) and modulation wheels, the octave transpose buttons, the controls for the dedicated vibrato LFO, fine tuning, and a knob that detunes both VCO2s simultaneously for instant fatness. (It also hosts the headphones output with its associated gain control, which is much better than finding the socket around the back.) Despite all of these on‑board performance capabilities, and despite the fact that its MIDI specification isn’t as advanced as that of the stand‑alone SEM, I particularly liked playing the TVS Pro over MIDI. I connected it to a 76‑note Roland workstation so that I was able to split SEM A and SEM B halfway up its keyboard, and play both (or control their sequences, if running) over a range of three octaves each rather than three octaves in total, and this opened up all manner of additional possibilities. I had huge fun controlling it using the workstation, running a sequence on SEM

Sample & Hold The TVS Pro’s Sample & Hold generator acts as a third source of control voltages, the other two being the keyboard and sequencer. Internally, this is routed to the oscillators’ pitches although you can direct its output elsewhere using patch cables. It derives its clock from the sequencer which means that, for example,

you can run one SEM from a sequence while obtaining random but synchronised effects from the other. Interestingly, this timing architecture means that you can also obtain ratcheted S&H effects, which is not something I can remember seeing on any vintage synth. Somebody is going to find a good use for this.

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A while playing a solo line on SEM B while routing the Roland’s Mellotron patches (which were also being sequenced by the TVS Pro) back through both of the SEMs’ filters. It was instant 1974... Phaedra and Rubicon had never been this easy! The one drawback, for me, was that the TVS Pro doesn’t understand MIDI sustain, which would have been useful. I also had fun directing the output of SEM A through SEM B, and then cross‑patching their audio and CVs to apply both LFOs to each of their sounds, or taking the audio frequency signals from one to cross‑modulate the other, or both... and much more. The results could range from exquisite to cacophonous. But since this was taking me well into modular synth territory, I soon ran up against the absence of any patchable mixers or VCAs in the TVS Pro itself. So it was now time to stick it in front of a true modular synth, connect its rear‑panel CV and Gate outputs to something sensible, route external audio back into its audio signal inputs, and see how it worked as a combination of controller, sequencer, sound source and external signal processor. It was superb; the marriage of the SEMs with a selection of Analogue Systems’ Moog and EMS modules was many a vintage synth enthusiast’s dream. Throw in a couple of Moogerfoogers and modern ambience effects and, well, you get the picture. My final experiment was to create some patches inspired by Roland’s LA Synthesis, wherein SEM A created the attack portion of the sound and SEM B provided the sustain and release. (The ‘chiff’ of a flute followed by its sustain is a good example of this.) With careful programming, paying

particular attention to the crossfade between the two sounds, the results could be excellent, although there were some sounds that SEMs wouldn’t generate, no matter how hard I tried. I would have loved to have had the time to sample the successful patches that I created for later conversion into a library of polyphonic pads — an enhanced 8‑Voice via the back door, if you like.

Shortcomings No synth is perfect, and the TVS Pro is no exception, although most of its shortcomings are minor or can be overcome. For example, I was surprised to find that, unlike the stand‑alone MIDI‑equipped SEM, the TVS Pro doesn’t understand MIDI CCs. I was also surprised to find that the keyboard isn’t polyphonic over MIDI. Conversely, I wasn’t surprised to find that the TVS Pro is slightly noisier than some other modern instruments. You can hear this as a quiet whistle if you switch off both oscillators and open the VCA. In the vast majority of cases this won’t be a problem, but it’s worth noting nonetheless. But here’s the biggie: Oberheim have tried to justify the TVS Pro’s lack of patch memories by saying that it’s an instrument designed to be played using its knobs and switches as well as its keyboard. I don’t buy into this. If I’ve spent hours refining sounds that work just so with each other and the sequencer, I would like to be able to save and recall them, whether tomorrow morning to continue writing and recording, or three years hence in front of 61,000 adoring fans at the opening ceremony of the new stadium at White Hart Lane. It would have been

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more honest to say that adding memories would have added considerable cost and complexity to what was already going to be an expensive instrument, and leave it at that. Either way, it’s a shame. I realise that I can’t save patches on my Minimoog or ARP 2600 either but, even as early as 1977, players could save most of the patch parameters on Oberheim’s OB1 monosynth and, nowadays, there’s no reason why you can’t combine top‑quality analogue gorgeousness with extensive memories. Given what’s possible with the TVS Pro, and what strange and wonderful sounds you may discover one day, only to be unable to recreate them the next, you may find yourself swearing quite a lot. Finally, it’s worth noting that the manual is rather brief and doesn’t tell you everything that you want to know. For example, it doesn’t seem to mention that aftertouch is transmitted over MIDI, and I had to experiment to find how to move the keyboard Split point. I admit that it’s clear and nicely presented, but it would benefit from being a bit longer and more encompassing.

Conclusions The TVS Pro is not a low‑cost synth, but that shouldn’t surprise you. The stand‑alone SEM Pro weighs in at a hefty $1200 or thereabouts so, if you start with something similar to two of these, add a mini‑sequencer and then place them all in a case containing a velocity‑ and pressure‑sensitive keyboard, it’s inevitable that the cost will be significant. But synths based upon SEMs have always been popular, and they command prices that cannot be justified using conventional

Alternatives There are a handful of third‑party modules based upon the SEM (the Analogue Solutions SEMblance springs to mind), almost all which are more affordable and extend the capabilities of the original design. It would be simple to take a pair of these and connect them to a suitable keyboard and sequencer to emulate the TVS Pro, or you could start with the Telemark‑K (essentially, a SEMblance with a keyboard) and build from there, although in all cases the results would lack the integrated neatness of the TVS Pro. Alternatively, you might be tempted to look toward newly conceived instruments such as the Moog Sub 37 and the DSI Pro 2. These are far from direct equivalents — the Moog is a dual‑oscillator monosynth that can allocate its two oscillators to the highest and lowest notes played, while the Pro 2 is a quad‑oscillator monosynth with a four‑voice paraphonic mode — and there are many other ways in which they differ from the TVS Pro and from each other. But all three are excellent instruments, so it would be worth spending a bit of time to see which best suits your needs.

price/performance criteria; people like the sound and the ergonomics and are willing to pay for them even when there are more affordable ways to achieve the same results. If you’re a fan of the Oberheim sound and you’re looking for an analogue synth for lead, bass and sequencing duties, you shouldn’t overlook the TVS Pro. I’m not sure that there will be enough buyers to justify its existence, but I sincerely hope that there will. The world needs labours of love like this and, if it proves to be successful, it will be fascinating to see what Tom does next. $$ $3495 WW

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Keith McMillen

Instruments K‑Mix Digital Mixer, Audio Interface & DAW Controller

The K‑Mix squeezes a mixer, an audio interface and a MIDI controller into one very tactile box. BOB THOMAS


ardware and software developer Keith McMillen is nothing if not inventive, and has a track record of over 35 years of musical innovation. In more recent years his eponymous company, California‑based Keith McMillen

Instruments (KMI), have introduced the SoftStep 3D foot controller, QuNeo 3D pad controller and the QuNexus and K‑Board keyboards, all of which feature interfaces based on KMI’s touch‑sensitive, ‘opto‑tactile’ Smart Fabric. The latest KMI product, the K‑Mix, builds on this technology to deliver not

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only an eight‑in/10‑out audio interface and DAW controller for Mac, iOS (very soon) and Windows (arriving soon, though no actual date as yet), but also an 8:2 fully programmable digital mixer, with three‑band semi‑parametric EQ plus dynamics on every input channel and the main output, three stereo aux sends and an on‑board global reverb. As a mixer, the K‑Mix can operate not only in conjunction with Mac, PC, iOS and MIDI, but also as a stand‑alone audio console in a live environment.

Three Into One Does Go Size‑wise, the K‑Mix is about the size of a large paperback. Its opto‑tactile control surface is made up of eight channel faders; one master fader; four circular control pads; nine channel‑select buttons; four buttons in a diamond‑shaped block that can either set the unit’s operating modes (‘banks’ in KMI‑speak) or act as MCU (Mackie Control Unit) transport controls; 16 buttons that give access to the functionality that you’d find on a similarly‑specified, full‑size digital console; a phantom power switch; and a preset recall button. Audio inputs and outputs reside on the rear edge of the K‑Mix. All eight analogue inputs are balanced, the first two being XLR/TRS combo connectors providing connection to KMI’s proprietary μPre microphone preamplifiers on the XLR and Hi‑Z instrument/line‑level sources on the jack. Individually switchable phantom power can be supplied to the XLR connectors at globally switched levels of 12V or 48V. The remaining six balanced TRS inputs can be switched to accept either line‑ or phono‑level signals.

Keith McMillen Instruments K‑Mix $579 PROS

• A highly‑portable, all‑in‑one, programmable digital console, audio interface and DAW controller. • High‑level sonic performance. • Innovative ‘opto‑tactile’ interface — no knobs or faders to wear out. • Further development and enhancement is underway and planned. CONS

• At this point in time, it is still a work in progress. • PC and iOS compatibility not yet available, but not far off. • Range of off‑the‑shelf DAW configurations not yet available. • Certain useful additional functions listed in the manual are not yet implemented. SUMMARY

The KMI K‑Mix is a one‑of‑a‑kind combination of programmable digital console, audio interface and DAW controller that has no moving parts, is light in weight and is smaller than a large paperback. What’s not to like?

Eight balanced TRS jack sockets carry the K‑Mix’s analogue outputs and are set up with the main output on 1+2, aux 1 on 3+4, aux 2 on 5+6 and aux 3 on 7+8. In surround mode, these can be configured in full‑range 8.0, 7.1, 5.1 and 4.0 modes. When the bass management feature is activated, all eight outputs are high‑passed and the low‑passed subwoofer information is sent out through the front‑edge mini‑jack

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headphone output. On the rear edge, in the space between input and output connector banks, you’ll find two USB ports. The ‘Audio’ micro USB port is used both to bus‑power the K‑Mix, and to connect it as an audio interface and/or DAW controller to an iOS device, Mac or PC. The ‘Control’ mini‑USB port connects either to a KMI MIDI Expander (equipped with physical MIDI sockets) or to the included power supply for use either when the K‑Mix is being used stand‑alone or when the Mac or iOS device to which it is connected cannot provide enough bus power.

Audio Interface In addition to its analogue audio functionality, the K‑Mix is a class‑compliant eight‑in/10‑out audio interface that is capable of supporting 44.1, 48, 88.2 and 96 kHz sampling rates. USB I/O setup is carried out in the editor and allows the K‑Mix’s eight inputs to be sent to the computer on an individual basis either immediately after the A‑D converter (Pre) or post EQ, dynamics and aux sends but pre‑fader (Post). On outputs 1+2 (the main outputs) the signal is sent from a pre‑fader point that is post the main output EQ, dynamics and return for the on‑board reverb, creating the possibility of having two sets of EQ and dynamics available on those outputs if required.

USB returns 1‑8 from the computer come back in on an individual pre/post basis. When the headphone socket is set to act as a discrete stereo output, the final two available returns (9+10) are routed to it, allowing the user to listen to, for example, a monitor mix from a DAW.

Control Surface When the K‑Mix is connected to a computer, three virtual MIDI ports are created, two of which (Audio Control and Control Surface) allow for direct two‑way communication between the K‑Mix and a computer or iOS tablet and, in the case of the Expander port, between K‑Mix and any connected MIDI device via the optional KMI Expander. How the control surface and ports operate is related to the K‑Mix’s operating modes or ‘banks’. The Mix bank is where all audio manipulation is carried out and, in this mode, the Audio Control and Expander ports are active. In this mode the K‑Mix does not output MIDI signals, with the exception of the ‘diamond’ pad which sends MCU‑compatible transport commands: start, stop, RTZ and record. Correctly configured incoming MIDI messages can control the Mix bank parameters — EQ and dynamics, aux send levels, pan, fader levels, etc — enabling the automation of the K‑Mix from a DAW or a connected MIDI device.

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Both the Control Surface and Expander are active in the three remaining banks: MIDI 1, 2 and 3. In a MIDI bank mode, the K‑Mix’s faders and ‘rotary’ pads can send MIDI CC numbers, and the switches can transmit any Note number on any MIDI channel except Omni. Individual switches can be set to act either as momentaries or as toggles. Controller and Note numbers can be set individually for each bank, giving the possibility of three sets of completely different CC and Note setups within each preset.

The Editor Although the K‑Mix can be operated without a computer, there are certain functions that are only accessible via the free K‑Mix Editor software. This editor has a neat, clutter free, multi‑screen interface that, in addition to those parameters that can be controlled from the K‑Mix itself, also allows access to additional functionality, both setup and operational. A complete K‑Mix parameter configuration, eg. for a particular track or song, can be saved via the Editor as a preset for instant transmission to the K‑Mix. Any number of presets can be stored and a selection of up to 12 of these can be downloaded to the K‑Mix for instant recall.

Mixing It Up As a standalone eight‑input/eight‑output digital mixer, the K‑Mix offers an impressive amount of on‑board functionality via its physical user interface. The faders and pads, although they require a definite though light touch, are very responsive, whether you are moving your finger along the surface or simply

tapping it to jump to a value. The position of the faders, the rotary controls being emulated on the circular pads, and the status of the switches, are indicated by the LEDs that can be seen through their semi‑translucent surfaces. At present, these LEDs are not particularly bright and their status can often be difficult to discern in strong light, an issue that KMI are actively addressing. To reduce friction between their surface and a user’s fingertips, the faders have a shallow, vertically‑ridged scoop in their surface in which a shallow horizontal notch indicates the unity gain point. For me it was the push‑buttons that took the most getting used to as, without tactile feedback, my initial tendency was to press too hard and I had to learn to ease off and to use a pressure that was only slightly greater than I’d use on a physical switch. Navigating around the K‑Mix’s surface is relatively simple, the fader and ‘rotary’ pad functionality being based on the chosen operating mode. For example, Main, Aux1, Aux2, Aux3 and input Trim are level‑control modes where only the faders are active. In Verb (reverb) mode the faders control the send levels from each channel and the rotaries act on all four available reverb settings, whereas with EQ, Gate, Compressor and Pan (which are essentially sub‑modes of the Main mode) the rotary pads allow you to access a core set of the full range of parameter controls that are present in the Editor, whilst the faders continue to act as channel faders. Fader and rotary resolution is more than high enough to satisfy most practical requirements. However, should you need more granular

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control, the Fine button gives you access to a significantly higher resolution +/‑ trim function with which to refine your settings.

In Use This review was carried out with the latest (V1.1 at the time of writing) firmware and Editor software (Mac only), which operated without any crashes or throwing up any bugs during the review process. However, certain global functions listed in the manual — Input Limiter, Backlight On/ Off, LED Brightness and Tone Generator — are not yet implemented. Putting these to one side, operating the

K‑Mix required very little in the way of a learning curve or specialist knowledge. The opto‑tactile control surface — touch it and lights come on or change colour — takes only a few minutes to become accustomed to and the white legending, tiny though it may be, is perfectly readable against its black background. Fader and rotary pad operation, once I had the required pressures stored in my fingers, was always reliable and smooth. Holding down the Shift button and touching the top or bottom of the channel fader activates Solo and Mute respectively. Multiple channels can be selected to either status, and these are

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indicated on the surface by changes in fader LED illumination levels and colour changes in the backlight colour of the channel select buttons. It is possible to put channels into Solo Mute status, and this is indicated by a flashing channel select button backlight. In VU mode, the fader LEDs show the signal level, and touching the fader switches the display, momentarily, to the fader position. Since touching a fader will snap its level to your finger position, the global Pass Thru fader function has to be activated to ensure that the level does not change until your finger passes through the current position. If you’re not

one for Pass Thru mode, you can simply toggle VU mode instead. The one aspect of K‑Mix’s physical user interface that took a little getting used to for me personally was the fact that its operational logic is based on assigning a channel to a function, making it impossible to assign a function to a channel. Once I’d changed from my usual ‘channel first, function second’ approach, this ceased to be an issue. Powerful and comprehensive though its physical control surface is, the K‑Mix really comes to life when being operated in conjunction with its Editor software, which gives you on‑screen access to all the

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K‑Mix’s functionality. The control surface is always able to control any on‑screen parameters to which it has access, whilst the Editor adds simultaneous on‑screen control of all these plus additional functionality and control, the combination allowing you to work extremely efficiently. In addition to all its DSP and control capabilities, the K‑Mix also sounds good. The KMI μPre microphone preamplifiers performed extremely well with both dynamic and condenser microphones on a variety of acoustic sources, and the line inputs dealt happily with an electric piano, a guitar preamp and a drum machine. The swept centre frequencies on the treble and bass EQ, in conjunction with the fully parametric mid range, make for a very effective and musical EQ. While the compressor and gate worked effectively, I wasn’t personally all that impressed by the on‑board reverb, which I felt would be fine for live use, but wouldn’t quite cut it in the studio despite the flexibility offered by its controls.

Conclusion KMI’s K‑Mix packs a surprising amount of functionality into its diminutive dimensions and, given the enhancements that are currently in the pipeline, it would seem that there’s a lot more to come. Space has restricted the level of operational detail that I’ve been able to go into in this review, so I’d recommend a close reading of the manual (available on the KMI web site) to fully appreciate the K‑Mix’s capabilities. The scope of KMI’s ambition for the K‑Mix means that it is currently a work in progress, in which the essential core functionalities of a digital mixing console,

Alternatives I can’t think of any faderless, knobless, mixing console/audio interface/DAW controller with touch‑sensitive control capabilities that is directly comparable to the K‑Mix at its price. It really is one of a kind.

audio interface and DAW controller are firmly in place alongside many of the operational enhancements that smooth and accelerate workflows, but where there is still much to come. As it is, with the current v1.1 Editor and firmware, the K‑Mix is an extremely usable, compact console that would be equally at home at the centre of a Mac‑based DAW setup, as a PA mixer for a solo artist, duo or small band, or as a submixer in a keyboard or modular synth setup. In the near future, once iOS and PC compatibility is released, functionality is enhanced and mapping becomes available for the major DAW platforms, I have no doubt that the K‑Mix will become a very attractive option for many musicians. I’m not normally an early adopter, but the K‑Mix’s combination of capabilities (current and planned) ticks so many boxes for me in terms of what I do and how I work that this one is staying here. If you’re in the market for a small programmable digital console, an audio interface and a DAW controller, the K‑Mix offers an attractive value proposition that you should certainly consider, especially once iOS and PC compatibility comes on‑line. $$ $579 WW

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SHADOW HILLS MASTERING COMPRESSOR In a league of its own, the celebrated Mastering Compressor is truly the master of compressors

Based on the EQP-1A3, the EQM-1S3 mastering EQ advances the recording process with even more high band frequency options

NEUMANN U47 FET COLLECTOR’S EDITION From kick drum to vocals and everything in between, allow the U 47 fet to bring its signature sound to your recording


Positive Grid Pro Series Compressor Modelling Plug-ins

Positive Grid’s latest plug-ins let you craft your own boutique compressor designs without ever touching a soldering iron. JOHN WALDEN


longside equalisation, compression is perhaps the most widely used processing tool within the mixing and mastering processes. It’s hardly surprising, therefore, that it has been discussed many times in the pages of SOS; and back in the September 2009 issue (

php?Month=9&Year=2009), Mike Senior provided a pair of articles that form an excellent ‘Compression 101’ class. As Mike explains, compression has been implemented using a number of different circuit topologies including VCA, vari-mu, optical and FET technology, as well as hybrid designs. Each approach has its own sonic characteristics, and these differences have led experienced mix engineers to

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prefer specific compressor designs for particular mix tasks. Of course, we now have the software compressor and, as computer power has advanced, so has the accuracy with which a plug-in compressor can not only control the dynamics of an input signal, but do so in a way that emulates the specific characteristics of classic hardware models. Waves’ CLA Classic Compressors Collection is an obvious example: it includes emulations of the FET-based UREI 1176, which is great for heavy compression on drums, guitars and bass, and the hybrid electro/ optical/tube Teletronix LA‑2A, which is famed for its silky-smooth sound, but can also be overdriven very effectively. These types of plug-ins can sound very good indeed but, although they’re a lot more affordable than the original hardware, can nevertheless come with a hefty price tag — the aforementioned Waves collection is currently $599. The other obvious point about software emulations of specific hardware is that, naturally, they tend to share the limitations of the original hardware. That’s not a criticism — it’s those sonic characteristics and design quirks that gave the device its reputation in the first place — but there are times when you might wish for more versatility.

Mix & Match So, how would it be if, within the overall ethos of something like a tube or FET or optical design, you could build your own compressor? Well, now you can, thanks to Positive Grid’s new Pro Series plug-ins. At a fairly modest introductory price, the first of these provides three distinctive compressor plug-ins, emulating tube, optical and FET circuits with options for swapping in and

Positive Grid Pro Series Compressors $99 PROS

• Classy-sounding compression in three distinct flavours. • Component-level modelling brings additional flexibility. CONS

• Option to ‘hide’ the inner controls would be nice. • Can we have some documentation please? SUMMARY

However the component-level modelling has been performed, Positive Grid are doing something right here. This first Pro Series studio bundle delivers three compressors with a ‘classic’ sound but a modern and flexible twist.

out individual components, so that you can create as many different variations as you like. Positive Grid have implemented similar component-level modelling in their BIAS Amp package, reviewed in the January 2015 issue. Initially launched for iOS but then ported to the desktop, this distinguished itself from the guitar rig software crowd by providing guitar players with a virtual workshop within which they could build their own guitar amp design. You can pick your choice of tubes, preamp block, tone stack, transformer, and any number of other elements, to create the amp of your dreams. And very good it is too. While component-level modelling itself is not unique, the way Positive Grid presented it to the user in BIAS Amp took it to a new level of accessibility. Having established the

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principles, they are now bringing the same approach to other studio equipment, and the expectation is that this release will be the first of many. As shown in the various screenshots, the three separate plug-ins look different, and those with a hardware background might detect some visual hints towards the original inspirations; for example, the optical model borrows some of its visuals from the Teletronix LA‑2A. Installation follows a fairly painless download/serial number combination and, with VST, AU, RTAS and AAX formats all supported, compatibility across major DAWs and OS variants should be broad. I did most of my testing with Cubase Pro 8.5 and experienced no particular issues. In terms of workload, enabling a single instance of

each model added a modest couple of percent to the average CPU load within Cubase on my test system. Understandably, this is more noticeable than, for example, the standard dynamics processors bundled with Cubase, and perhaps just a little more than that generated by the similarly priced Waves VComp, for example. You might not want to use the Pro Series compressors on every track but, unless CPU cycles are very constrained, multiple instances on selected tracks ought to be possible.

Open Ended At one level, the Pro Series compressors are fairly standard, if ‘boutique’, compressor emulations; you get an appropriate set of front-panel controls for each of the different

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models, and vintage-style VU meters that can be toggled between displaying input, gain-reduction or output levels. The FET model presents the most comprehensive control set, with input, threshold, ratio, attack, release and output levels. The front-panel controls of the optical and tube models are simpler — there’s nothing labelled ‘ratio’, for example — just like those of their hardware inspirations. All three, however, offer additional control options ‘inside the case’. These are options that, in the original hardware, would have involved screwdrivers, soldering irons and a risk of electrocution to change. This component-level editing is most obvious within the tube and optical models, each of which offers four components that you can swap in and out based upon a number of selections. The FET model is less customisable in this regard, but all three plug-ins also have a series of additional controls positioned along the base of the virtual faceplate, representing further

tweaks the user could make to the internal circuitry. These include wet/dry options if you want to blend uncompressed and compressed signals for parallel processing, while the FET model includes low and high cut options, which, I think, allow you to EQ the side-chain signal so that compression is triggered by a specific frequency band. I say ‘I think’ because, aside from the fairly general description on Positive Grid’s web site, the plug-ins are not supplied with any supplementary documentation. At one level I can understand this: installation is easy and the process of adjusting the control set requires no explanation. However, at another level, the studio geek in me wanted an introduction to the three models, a commentary on why there are different, a background to the component modelling process and, most importantly, to be given some guidance (in the absence of knowing anything about the insides of a real hardware compressor) as to why I might, for example, choose a mica capacitor as

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opposed to a ceramic one. A basic PDF reference manual would not go amiss.

Get Stuck In I started my exploration with the optical compressor emulation, and I nearly didn���t get much further, because I have to say I got totally hooked in exploring just what this plug-in could do with a variety of different sources. I was almost universally impressed. Yes, you can turn a drum track into something lifeless if you’re careless but, equally, you can also go from punch to pump very easily depending upon what your track requires. And, like the LA-2A hardware that inspired it, Positive Grid’s optical model sounds particularly good with vocals. It’s smooth and warm and, if you dial in more gain, then you can add a very nice touch of tube drive. I could easily see this becoming a firm favourite for my own use. What was particularly noticeable was just how much gain reduction could be applied to a sound while remaining, if not transparent exactly, still very musical. I’m not quite so sure I understand what the different capacitor options really bring to the party, but switching the input tube types and the light source (bulb, LED or panel are the choices, and you can also adjust the age and

sensitivity of the light source) most certainly changed the character of the sound. This wasn’t always about the compression itself; there were also some noticeable tonal changes. While the additional controls on this model allow you to adjust the attack and release of the compression (again, it would be nice to have a manual to explain the roles of the two different release knobs), as well as gradually shift between hard- and soft-knee response, once you have set these and picked the component options, the main Gain and Peak Reduction controls are really all you need. From a screen real-estate perspective, with all three models, it would therefore be useful to have the option to fold away the ‘inside’ controls and just leave the front panel. Perhaps this is an option for a future update? I perhaps wasn’t quite so blown away by the (Fairlight-inspired?) tube model. When not pushed too hard, the compression worked beautifully and suited almost any source material. However, pushed a little harder, this emulation seemed to be more about adding ‘character’ rather than remaining transparent at high levels of gain reduction. That said, it you want to crunch up your drum bus, this plug-in has just the preset — and the fact that each of these plug-ins is distinct is, of course, a good thing. The FET model is possibly 1176-inspired, although you do get an expanded (modern) control set here. The low-cut and high-cut controls allow you to focus the compression detection on specific frequencies, for instance to prevent something like a kick drum from triggering exaggerated compression. When mixing, the combination of look‑ahead and fast attack times makes it useful for a diverse range of tasks. It sounds excellent on vocals, acoustic guitars, bass

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and drums, where you can really smash into the compressor if required, but the flexibility of the control set and the transparent sound also make it very useful as a mix-bus compressor. Of the three plug-ins, I felt this was perhaps the most transparent and flexible, delivering compression (and plenty of it if asked) while doing its best not to alter the tonality.

All In All With any emulations of classic gear, it is possible to have an almost endless conversation about how close the sound of the software is to that of the hardware. Up to a point, that is a conversation worth having but, at the classier end of the plug-in marketplace, I think the answer has been

Alternatives If you are happy to just get the emulations of classic hardware compressors, there are plenty of options that cover the same ground as the Pro Series. For example, Waves make some top-notch software emulations of all the ‘names’ mentioned here, but they are considerably more expensive than the Pro Series. At the other end of the scale, there are also plenty of freebie options, and some DAWs now ship with some more esoteric compressor plug-ins as part of their bundle. For example, Logic Pro X’s Compressor now offers some quite credible VCA, FET, optical and digital emulations. However, none of the competition provide the DIY workshop experience of the Pro Series and the ability to pick through certain component-level choices to get the compression sound you are after.

‘pretty close’ for some time. Perfect? Well, perhaps not, but I suspect Positive Grid’s Pro Series compressors actually do well in this regard. For many of us to whom the hardware is not an option, there is, of course, a more pragmatic discussion to be had that ignores the absolute accuracy of the model and instead considers whether they sound good in their own right. And, on this front, I think the Pro Series scores very well indeed. Of the three, I think I’ll find myself reaching for the optical and FET models most regularly; they sound great and, between them, can deliver smooth and transparent or punchy and characterful sounds. And, if you are lucky enough to pick them up at the special sale price ($99 at the time of writing), I think they represent a great deal. The component-level modelling brings additional flexibility to what’s on offer. While this is undoubtedly impressive — and a lot of fun to experiment with — I don’t think it is quite so fundamental to these plug-ins is it is with BIAS Amp. These are fine-sounding compressors with or without the option for changing the components. That said, I can’t help but wonder what might be next off the Positive Grid shelves in terms of component-level modelling, and I’m looking forward with some anticipation. However, whatever it is, can I please get it with a manual? $$ Sale price $99; full price $199. WW

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DMG Audio Limitless Mastering Limiter Plug-in

If you’re feeling limited by your existing limiter, perhaps DMG Audio have the answer? SAM INGLIS


ave Gamble is a plug-in developer who sets out to make his products the very last word in whatever it is they do. His Equilibrium equaliser and Compassion compressor are already the most comprehensive examples of those two processors I’ve ever come across, and now he’s turned his attention to the challenge of creating the ultimate plug-in mastering limiter. At first glance, this seems like a less ambitious goal than designing the mother

of all equalisers, or the compressor to end all compressors. After all, a mastering limiter is intended to do one very specific thing: to make your mixes as loud as possible, with as few side-effects as possible. Many limiting plug-ins thus have hardly any user controls beyond a simple threshold or gain setting — but you probably won’t be surprised to learn that DMG Audio’s Limitless is not among them.

Three Steps To Heaven Putting a simple output limiter across the master bus is fine when you need to send clients a quick reference mix, but mastering engineers in pursuit of the best results will often use more than one stage of processing. The reason for this is

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that dynamic variation within programme audio happens on different timescales. At the ‘micro’ level, most mixes contain instantaneous, transient peaks caused by events such as drum hits; but the level of the audio also changes in a ‘macro’ fashion too. To achieve the loudest possible master with the fewest possible side-effects, it may be necessary to tackle longer-term dynamic variation separately from the transient peaks. What’s more, many mastering engineers don’t only use limiting to control the latter: there are cases where allowing transients to clip the input of an A-D converter can actually sound more natural than having a limiter do all the work. Limitless reflects this approach and includes not one but three processing stages, all of them highly configurable. Two separate limiters are designed to work in tandem; the first is intended to allow

DMG Audio Limitless £150 PROS

• Extremely configurable, yet easy to use and immediate. • Its three-stage multiband processing can achieve impressive levels of transparent gain reduction. • Excellent graphical feedback. • Sensibly priced and not too CPU-intensive. CONS


Whether you want a good-sounding ‘set and forget’ limiter or a processor that allows you to dive in and fine-tune every last parameter, Limitless ticks all the boxes.

transient events to pass through, but control dynamic variation with slower attack and release characteristics. The second then squishes the transients, in conjunction with the third of Limitless’s processing elements: a soft-clip stage preceding the limiter, which can mimic the characteristics of several different clipping options. The limiting can be configured as a conventional full-bandwidth process, but Limitless also offers the option to have it operate independently in up to six frequency bands. This can help to achieve natural-sounding results with material that has loud peaks in specific frequency ranges, because you can ensure that other areas of the spectrum are not ducked along with the peaking frequencies.

Soft & GUI There are times when the sheer range of options available in Compassion or Equilibrium can feel overwhelming, but that’s not the case here. Although Limitless is easily the most comprehensively featured limiter I’ve ever come across, DMG have managed to harness all of its power within a friendly and well thought-out user interface. By default, Limitless opens in a fairly small window that presents only the main Threshold, Ceiling and Release time controls on the left, and the output meters on the right. However, the window can be freely resized, and clicking a small icon in the plug-in toolbar makes visible a list of additional parameters in the lower left and right panes. The large central section, meanwhile, is devoted to visualising the settings of the band crossovers and the effect of any limiting on the input signal. The default visualisation shows an FFT-style

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“Though you don’t have to use it, the multiband option can be really effective when you need more level with fewer side-effects.” These two basic alternatives complement each other nicely: the frequency view gives you a clear idea of how the energy within your mix is distributed across the spectrum, and how the limiter is behaving in each band, while the time view lets you pinpoint how much limiting is taking place at any given moment. And if your main concern is to hit a particular peak loudness value, another alternative visualisation supplements the numerical LUFS readout below the main output meters with a scrolling histogram. The behaviour of all of these displays is highly configurable, thanks to a range of global and instance-only preferences, accessed from the Setup button.


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instantaneous plot of peak level across the frequency spectrum; when the limiter bands are applying gain reduction, the top part of the graph turns a lighter shade of blue. On this is superimposed a fairly conventional EQ-like interface which allows you to configure the band splitting. A simple click enables and disables the bands, while clicking and dragging adjusts the gain and centre frequency of each (though this behaviour can be customised). If you so choose, this frequency view can be replaced by a neat scrolling waveform display that can be sync’ed to song tempo, with limiter activity displayed in red and green above and below the programme audio.

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Don’t Cramp My Style Limitless is not the first limiter I’ve used that offers different ‘styles’ of limiting, with names such as ‘punchy’, ‘transparent’, ‘aggressive’ and ‘smooth’. What is new, at least to me, is the extent to which the intrepid user can dive in and adjust the various parameters that make up a style. When you select one of Limitless’s preset styles, only four additional ‘expert’ parameters are visible in the expanded interface, but if you choose the ‘manual’

Alternatives There are already many excellent limiting plug-ins on the market, though I don’t know of any that are quite as configurable as Limitless. Alternatives worth investigating include Waves’ L3-16, FabFilter’s Pro-L, Sonnox’s Oxford Limiter, Slate Digital’s FG-X and IK Multimedia’s Stealth Limiter.

style, or copy one of the preset styles so that it can be edited as a manual style, you get the full list of Advanced controls. These include such factors as lookahead, knee, ‘weighting’ — which sets how gain reduction is distributed between different frequency bands — release ‘shape’ and finally Dynamics, which controls how much of the work should be done by the transient limiter and how much by the peak limiter. Engage Clipping on the right-hand side of the interface, and here, too, you’ll be presented with plenty of control over the process. Three different flavours of clipping are on offer; the two ’swell’ options are described as “simple waveshapers which mostly add third-harmonic distortion to increase perceived level”, while ‘knee’ offers hard converter-style clipping at one end of the spectrum and smoother soft clipping at the other. Reducing the Amount control from 100 percent lets you mix in some of

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the dry, unclipped signal, and there are also Drive and Trim controls.

No No, No No There’s No Limits In practice, I found Limitless’s multi-level interface very well thought-out. Thanks to the simple default view, you can be up and limiting within seconds of installing it, and the results are good enough that I can imagine many users never needing to take things further. But when you do delve deeper, you quickly begin to get a feel for which styles of limiting suit different types of programme material; and when you go further still, you soon start to understand which controls are key in creating your own custom settings. Though you don’t have to use it, the multiband option can be really effective when you need more level with fewer side-effects. I’ve used similar features before in plug-ins like Waves’ L3, but what was really a revelation to me in Limitless was the clipping. I’m sure most of us have found that saturation or ‘analogue warmth’ plug-ins on the master bus can give a welcome increase in apparent loudness without bringing up the peak level, and you can achieve something of the sort here

using the softer clipping options, but what surprised me was how hard you can push the clipping in ‘knee’ mode without audible side-effects. When a plug-in sounds great and is absurdly comprehensive, yet easy to use, you have to dig pretty hard to find anything to complain about, and I haven’t even mentioned the many little touches that help to elevate Limitless above the herd. There is, for instance, an excellent PDF manual, while features like the built-in high-pass filter, optional inter-sample peak detection, constant-gain monitoring and very flexible dither noise shaping are all welcome if you need them and easy to ignore if you don’t. All in all, I can’t recommend Limitless highly enough. Not only is it immensely flexible and capable of a lot of very transparent gain reduction, it’s also more affordable than many alternatives, and surprisingly economical on CPU load. Limitless has already become my first-choice output limiter, and it’ll be interesting to see if anything else out there can top it. $$ £149.99 (approx $212) WW

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IGS Audio Springtime Four-channel Spring Reverb We get hands-on with the most unusual device yet to spring forth from the IGS workshop. NEIL ROGERS


n a world where more and more mix processing is done inside the computer, a large, rackmount, all-analogue four-channel spring reverb is a pretty bold product to bring to market! Unlike, say, a preamp, EQ or compressor, it’s not something you’d normally go out of your way to use when tracking most sources, and when it comes to mixing there are plenty of high-quality in-the-box reverbs — and for this money some serious outboard processors too.

So when I was invited to review IGS Audio’s not-exactly-budget Springtime I was intrigued: what could it offer that conventional springs or software emulations could not, and how best could I incorporate the formidable-looking unit into my current workflow?

The Long & The Short With its imposing 3U 19-inch rackmount form factor, plentiful large, solid-looking knobs and pleasingly large VU meter, the Springtime looks striking. It feels solid too — it gives you the definite impression

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of no-compromise quality. Despite the number of controls, it’s laid out nicely, and everything’s thus very easy to digest. Inside are two pairs of physical springs, one ‘short’ and the other ‘long’. Each is fully independent of the others, so each pair can be configured as one stereo channel or two mono ones. You could have up to four separate mono reverbs, then, and, as each channel has its own balanced input and output on XLR, you could also choose to cascade the springs. The four channels all have their own dry and wet controls, hard bypass and, impressively, a bypassable three-band EQ for the wet signal, with Low, Mid and High controls to allow you to ‘focus’ the reverb sound as you wish. All channels share that big VU meter, and there’s a chicken-head switch to choose which feeds the meter. When contemplating how I might work with the Springtime in my own studio, it dawned on me that the vintage AKG BX5 and Great British Spring reverb units we have here at Half-Ton Studios haven’t been getting much exercise of late, even though I generally enjoy the characterful results when I  do dig them out. Usually, I’ve turned to those devices for a mono and distinctly lo-fi effect, but the Springtime is too well-crafted and too serious an investment to be merely an occasional character piece. With that in mind, I decided to put it through its paces as a main mix reverb, before exploring how it might add further value in other ways, such as during tracking, or for more experimental effects. When I took receipt of the review unit, I had a mix project on the go which had a minimal Talking Heads-style vibe. It seemed ideal for a bit of spring-reverb

IGS Audio Springtime $3195 PROS

• Two variations of rich, warm-sounding spring reverb. • Individual three‑band EQ for all channels. • Great build quality and styling. • The feature set does offer some nice flexibility around the theme. CONS

• Not cheap. • A niche effect perhaps — with limited use on some material. SUMMARY

Despite the hefty price tag, IGS Audio have produced a very high-quality analogue effects unit that would enhance many sources with it’s deep, rich, but unmistakably spring reverb sound. Two different flavours of spring are included and there are four channels available to be used individually or as stereo pairs.

treatment, so I patched the Springtime in and configured it to operate as two stereo hardware sends in Pro Tools. (When using the unit like this, you need to set the wet control all the way up and the dry all the way down.) Using primarily the short reverb in this track, it was a lovely moment when I first heard it applied to a sax part; it immediately added a sense of space, depth and fullness that required very little messing around with to sound, well, fantastic — so I swiftly dished the same effect out to other elements in the mix. Despite the effect not being as obviously transformative for other sources, it certainly added a richness that seemed

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to work perfectly for this song. Listening to the snare track in solo, it had that classic ‘boing’ that only spring reverb can give you. It sounded great when used subtly on this song and I think this effect would excel on a snare in a sparse dub or reggae track. Playing with the EQ controls on the short channel, I found I had to get quite ‘stuck in’ for it to have a really noticeable impact on the tone. That said, it was great to have the option of seemingly being able to use more of the effect by dialling back the top and low end, and towards the later stages of the mix I found myself fine-tuning the EQ rather more. The difference between the short and long springs wasn’t as dramatic as I’d anticipated, with the longer spring offering an ever-so-slightly longer decay. It also has a slightly darker sound, and a slightly different character that’s difficult to articulate. This long spring seemed to really work well on sources that benefited from a little thickening; a female vocal,

for instance, or a clean electric guitar part. It was while using the long reverb on vocals that I felt I really began to extract some value from the EQ. For example, it felt like I was shaping much more than the tone of the effect when I pushed the mid out and dialled back the ‘high’ frequencies a little on a soft female vocal track.

More Reasons To Spring? Over the review period, I got the chance to use the Springtime in a few different recording and mixing scenarios. I took the opportunity to experiment with it as an insert effect, mainly trying to incorporate the reverb along with some additional outboard mix processing I was using on a few different styles of vocal. When used as an insert, you set the dry control to 100 percent and blend in your desired level of effect with the wet control. For me, the latter typically ended up between 10 and 20 percent. I created a really nice effect using the

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long spring on a female rock vocal and had fun increasing the wet level in the choruses. Spring reverbs and guitars are the firmest of friends, of course (they’re often tracked with an amp’s spring reverb engaged) and running some clean guitars through a combination of long and short reverbs in mono worked a treat. While stereo reverbs can often sound lusher, it’s really refreshing to use a mono reverb sometimes, as it means you can apply it very generously and then place it where you want in the mix, perhaps in the same place as the source or perhaps panned opposite the source signal if you want stereo width.

“Not only does it sound unique and wonderful, but it forced me to reconsider some aspects of my mixing workflow.” I also used the Springtime on a guitar tracking session, for which I had the reverb set up such that we were monitoring with the reverb engaged, but recording the dry and wet signals to separate channels, to give some room for manoeuvre in the mix. Did it sound any better than the reverb that was on the Fender amp we were recording? These things are highly subjective of course, but the guitarist certainly enjoyed the results and it seemed to give a great deal more richness to my ears, not to mention greater control over the result, courtesy of the onboard EQ.

On their web site, IGS suggest trying the unit on a whole mix, and their examples seem to work really well, but this tactic seemed a little heavy handed for the tracks I had to hand. The fact that I’d have been quite happy to run a whole mix through the unit, however, is testament to its high sonic quality, and when the reverb is bypassed, the sound remains immaculate. Oh, and it also does that very cool thing that spring reverbs do if you give it a gentle slap (don’t worry, IGS, I was very careful!); as the spring vibrates it can create some quite dramatic effects.

Reverb Rated I thoroughly enjoyed using the IGS Audio Springtime and when I employed it as a main reverb on the right material it sounded nothing short of fantastic. But it offers plenty of other creative applications, both when mixing and when recording, and the choice of multiple mono or stereo channels, combined with the onboard EQ, offers flexibility. Like many people these days I mix primarily in the box, so unique and high-quality gear such as this poses something of a dilemma: after the honeymoon period I always enjoy with new toys, would I really continue to use it? I’m not entirely certain of the answer, but using the unit during a few mixes has reminded me that mixing entirely in the box can breed a little laziness, and that it’s really not that hard to incorporate a few pieces of choice equipment into my workflow. And I remain convinced that a few of the tracks I worked on came together a little more quickly and easily than they usually would, which has given me food for thought.

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I performed a few quick A/B comparisons against some spring-reverb plug-ins I use regularly, and I was left in little doubt that the hardware offered a greater sense of depth and space. Perhaps this difference would be a lot less noticeable on consumer playback systems. But on the other hand, now we’re all using similar effect plug-ins, maybe several such small differences could add up to create a very big one? The elephant in the room, of course, is the price. The Springtime is a superb piece of equipment that oozes quality and sounds immaculate, but all this comes at a significant price — and I suspect that it could prove difficult to persuade many people to part with such a chunk of their hard-earned cash for as quirky a device as this. When making significant gear purchases for the studio, I always have one eye on the likely future resale price, and that’s one reason I often choose ‘heavy-hitter’ brand names — I know that if I change my mind or end up using it less than I’d envisaged (or need to free up funds to invest in a crazy spring reverb...) I’ll be able to recoup a lot of my initial outlay. This unit’s likely depreciation is quite tricky to gauge, although the quality and the fact that vintage spring reverbs command quite a decent price means you should get a fair chunk back. It occurred to me that there may have been a few design options that could bring the price down, or result in more for your money. I’d certainly be intrigued by a single stereo/dual-mono version with only one spring-length option, and maybe even without the EQ (most people contemplating buying this will have other

Alternatives There’s nothing quite like the Springtime. That said, vintage AKG spring reverb units such as the BX10, BX15 and BX20 can offer a nice vibe, as could a vintage Roland RE301. Plenty of guitar amps and tonewheel organs feature spring reverbs, of course, and there are plenty of stompbox and software simulations. But in terms of currently available genuine spring reverbs, you could check out the Vermona ReTubeVerb Tube Spring Reverb, the Furman RV-1 and the Demeter RV-1D Real Spring Reverb.

EQs available). Alternatively, if the EQ could be made accessible separately from the reverb, that might increase its utility. But, given that you need a certain amount of physical space to house real springs, going ‘all in’ like this does make a certain sort of sense from a design point of view. Only smaller manufacturers such as IGS have the creative license to be able to stick their neck out with a unique high-cost, low-volume product such as this, and I think it’s wonderful that they’ve made it happen.

Final Thoughts The high price doesn’t make the Springtime poor value for money. My initial reaction that the whole thing was, frankly, a little bonkers changed when I investigated its charms in detail. Not only does it sound unique and wonderful, but it forced me to reconsider some aspects of my mixing workflow. It left me wondering if, where the material is suitable, such a device could have a greater positive influence over my work than, say, spending a similar amount on new software effects and processors.

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Audio Examples For each source, I have included a dry example along with ‘short’ and ‘long’ spring sample options. I’ve been deliberately generous with the amount of reverb in order to help illustrate the differences. The audio filenames should give you a hefty clue as to what each one contains! Acoustic GTR Dry

Electric GTR long

Acoustic GTR Short

Sax Dry

Acoustic GTR Long

Sax Short

Drums Dry

Sax Long

Drums Short

Vox Dry

Drums long

Vox Short

Electric GTR Dry

Vox long

Electric GTR Short

Spring reverb certainly isn’t an effect that works on every mix, but I enjoyed using the IGS Springtime. It’s a well built, great-sounding and versatile spring reverb and while the price will be an issue for many people, IGS may just have identified a large enough niche market to make it viable. In case you’re curious about the sound, I’ve supplied a number of audio examples, which you can find on the SOS web site (, but

if you can find somewhere to audition the IGS Springtime on your own material I’d suggest you grasp that opportunity with both hands. If you already work with outboard or material where spring reverb works particularly well, and have the funds, I can highly recommend it. $$ $3195. TT IGS Audio +48 601 597 592 EE WW

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WIN! Focusrite F

recording bundle

ocusrite sprang into existence in 1985, when Sir George Martin asked legendary console designer Rupert Neve to create a new series of modules for London’s AIR Studios. Among them was the ISA 110 — short for Input Signal Amplifier — an excellent microphone preamplifier. The mic preamp in today’s Focusrite ISA range is almost identical to the original design, and the ISA 828, part of this prize package, provides eight of these classic mic preamps. Each of these includes selectable input impedance to suit any microphone, classic or modern, and the whole arrangement fits into a single 2U 19-inch rack module. Focusrite are also including the optional eight-channel, 192kHz A-D converter card for complete digital audio compatibility. This means eight-channels of single/dual‑wire AES3 and S/PDIF, as well as an eight-channel ADAT lightpipe (SMUX/SMUX2) output. Focusrite’s latest interface designs remain influenced by the ISA Range, and the new Red 4Pre included in this



A distinctive feature of the Red 4Pre is its use of the Dante AOPI (audio‑over‑IP) networking technology, which enables you to connect up to 64 additional channels to the Red 4Pre using Focusrite Red or RedNet modules — or indeed any Dantecompatible device — virtually anywhere there’s an Ethernet connection. Already widely used in live sound and educational facilities, Dante allows up to 512 channels to be carried via a single Ethernet cable. You can place inputs and outputs exactly where they’re needed and you can forget cumbersome and expensive multicore cables. You’ll be able to use

unique Focusrite package exemplifies the powerful combination of a traditional approach to high-quality sound coupled with the very latest technology. The pinnacle of Focusrite’s interface range, the Red 4Pre, with up to 58 inputs and 64 outputs, combines four of Focusrite’s new Red Evolution mic preamps with dual Thunderbolt 2 and direct Pro Tools HD connection, alongside Dante network audio connectivity. In addition, the Red 4Pre offers latency levels so low that you can record with your favourite DAW plug‑ins and virtual instruments with no need for external DSP or cue-mixing utilities. Each software-controlled preamp includes the unique ‘Air’ effect that recreates, in the analogue domain, the characteristics of the ISA preamplifier. The preamps also boast an impressive noise performance of -129dB EIN, and also up to 63dB of gain. Featuring parallel‑path summing conversion for maximum dynamic range, and a beautiful yet robust design, the Red 4Pre interface is the perfect balance of form and function, delivering the sound quality and versatility engineers and producers expect from Focusrite. As SOS reviewer Sam Inglis put it in his May 2016 review (http://sosm. ag/may16red4pre), “Focusrite’s ‘best interface ever’ provides impeccable sound quality, [and] few can match its potential for expansion.”

Dante straight away, as rounding out this package is the new RedNet AM2 stereo headphone amplifier and line output interface. RedNet is Focusrite’s flagship Dante interface range, and the AM2 offers a high-quality Dante headphone/ loudspeaker stereo monitoring solution with superb digital to analogue conversion at up to 96kHz, providing transparent and accurate audio. The powerful built-in headphone amp delivers plenty of level — even with high-impedance headphones — for maximum high-quality audio, and there are independent headphone and line output level controls. On top of that, an integrated Gigabit Ethernet switch lets you connect a second device for simpler cabling. This extensive prize package spans the history of Focusrite’s dedication to quality sound, combining classic microphone preamplifiers heard on hits around the world with the very latest in Thunderbolt and Ethernet-based audio. To be in with a chance of winning this spectacular bundle, follow the link and fill in the form on our web site before Monday 8 August, 2016. Good luck! Prizes kindly donated by Focusrite. TT +1 310 322 5500 WW

TO ENTER, PLEASE VISIT: July 2016 / w w w . s o u n d o n s o u n d . c o m


do you do what you do? “I have been emotionally and spiritually compelled to work with music since I was 5 years old. Listening to music, playing a few instruments and singing all led to my professional career as a producer, engineer and music mixer.”


does Aurora help you do what you do?

Ron Saint Germain

Saint’s Place Studio credits: Jimi Hendrix, 311, Living Colour, Nels Cline, Whitney Houston, Soundgarden, Creed and many more.

photo • Karsten Staiger

“The Lynx Aurora 16 sounds amazing. I truly appreciate that they do not add any low frequency ‘bump’ or ‘polish on the top’ to make you THINK your work sounds better than it really is. Their sonic performance is truly spectacular, as is their unrivaled customer support.”

For over four decades, Ron Saint Germain has engineered, produced and mixed an eclectic range of performances, from Jimi Hendrix to 311 to Living Colour to Soundgarden, garnering 14 Grammy’s® for the artists along the way. His converter of choice since 2008 at Saint’s Place in northern New Jersey is a rack of Aurora 16HD converters. Aurora’s clarity and transparent, open audio quality are a perfect match for his Neve Amek 9098i 128-channel mixing console and extensive vintage analog signal processing that are essential for Ron’s exacting audio requirements. To see and hear more about Ron Saint Germain and Saint’s Place Studio, go to the LynxStudio YouTube channel.

converting the masters of sound

©2014 Lynx Studio Technology. Aurora is a trademark of Lynx Studio Technology


Phaedrus Audio Phamulus Mono Vari-mu Compressor

Can a tiny box like this really deliver the classic vari‑mu sound for so little money? BOB THOMAS


aidstone-based manufacturers Phaedrus Audio make a growing range of hand-built tube and solid-state audio equipment. When I started my research on the brand, I discovered that the designer of the company’s London series of (mostly) tube-based products — “inspired by the leading British studios of the late 1950s”,

as Hugh Robjohns put it in his review of the Hydra mic preamp in SOS March 2016 — is none other than Richard Brice, a name that I have encountered over the years in a variety of roles, ranging from hi-fi journalist and broadcast equipment designer to developer of the Francinstien stereo enhancement system. He is also the author of several fascinating books, including Multimedia & Virtual Reality Engineering, Music Engineering

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and Music Electronics. What I hadn’t appreciated until now was not only the extent of Brice’s career in the commercial side of broadcast technology, but also the depth of his musical background, which, no doubt, has contributed to his design work at Phaedrus Audio.

The Box As Hugh noted in his review, all of Phaedrus’ London series units share an unusual blue-bezelled, extruded-aluminium compact form factor that allows them to be used as desktop devices or, using an adaptor plate, to be rackmounted in pairs. Despite this forewarning, the Phamulus’s diminutive dimensions were a bit of a surprise, accustomed as I am to finding vari-mu compressors housed in multi-U 19-inch rackmount enclosures. Phaedrus make no secret of the fact that the Phamulus is based on the circuitry of the Altec Lansing 436C. There have been various ‘reworkings’ of Altec’s basic design, the most famous and extensive — and perhaps also the most revered — being EMI’s modification of the 436B in 1960. With the addition of a matched-impedance input-level control, a six-position release-time switch (with a unique hold function between each setting), an output-level attenuator and a new front panel (among other changes), the 436B turned into the RS124 that saw service in Abbey Road Studios, most famously with the Beatles. But Joe Meek also modified Altec units in his Holloway Road Studios, and various others similarly repurposed what were at the time relatively affordable compressors. The most substantial alterations

Phaedrus Audio Phamulus £599 PROS

• Excellent compressor/limiter performance. • Great sound. • Compact. • Serious value for money CONS

• Fixed attack. • No side-chain filter. SUMMARY

Essentially an Altec 436C in a small box, the Phaedrus Audio Phamulus delivers the core of the performance of that classic variable-mu compressor at a very attractive price.

(apart from physical size) that Phaedrus have made are the use of a solid-state side-chain, the addition of a 12-position, detented front-panel control of the unit’s release time, a switchable -10dB output attenuator and the 600Ω internal load resistor that ensures compatibility with present-day equipment. Alongside the input‑level and release-time rotary controls, the Phamulus’ front panel carries an In/Out switch for the fixed -10dB output attenuation, and a Compress/Limit selector. Being based on the 436C, the Phamulus carries that unit’s front-panel variable threshold trim pot, which is active only when the Limit option is selected. In Compress mode, the Phamulus’ threshold is set at 0dBm, which gives a compression ratio of 2:1, whilst the Limit threshold is factory set to the

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maximum -16dBm, producing a 4:1 ratio at that level. Increasing the Limit threshold via the front‑panel trim pot will result in compression ratios between those two extremes. The final front-panel feature is the small, circular VU meter that displays the amount of gain reduction taking place, and the trim pot used to zero the meter if necessary, should components drift in value or be changed.

“Where the Phamulus really worked for me was sitting after the preamp, compressing a vocal, acoustic guitar or bass as it was being recorded.” On the back panel you’ll find input and output balanced XLR connectors, the jack for the stereo link cable, and the power-supply connector.

Circuits Although based on the 436C, the Phamulus isn’t a slavish point-to-point wired copy, but it does carry over that which Phaedrus describe as the “natural contour” of hook-up wire in the curved tracks of its PCB boards, and the “classic” practice of star-earth grounding. As in the 436C, the balanced incoming signal passes through the unit’s input transformer and input attenuator to the Phamulus’s variable-mu element; a twin-triode, medium-mu, semi-remote cut-off 6BC8 or 6BZ8. These tubes were originally developed for consumer television receivers in the days of CRT screens and, as a result, are readily


available at low cost as NOS (new old stock). All the voltages necessary internally are derived from the 12V AC wall-wart power supply that, to Phaedrus Audio’s eternal credit, is supplied with a locking connector. An alternative and vastly more expensive external power supply, the Phuel, is capable of supplying two London series components and is available for those with deep pockets and exquisite sensitivities. In the Phamulus, the solid-state side chain that controls the gain of the 6BC8/6BZ8 is based on a double-diode setup that replaces the 436C’s 6AL5 tube. The side-chain preset release-time trim pot in the 436C is replaced by the front-panel Release control, whilst the threshold control remains as a trim pot acting on the bias of the double diode, rather than on that of a 6AL5 tube. From the 6BC8/6BZ8, the balanced signal travels to the output amplifier, a 12AU7/ECC82 twin-triode valve in a push-pull configuration, and onwards,

through the output transformer and switchable -10dB attenuator, to the 600Ω load and then to the outside world. You may have noticed my lack of mention of a hard (or soft) bypass switch, and that is because there isn’t one — the Phamulus is always in circuit. Personally, I can live without a hard bypass, but I’d have liked there to have been a soft-bypass switch that disabled the side-chain, thereby turning the Phamulus into a line amplifier with 30dB of gain for that overdriven transformer sound that can sound so delicious on the right source. Incidentally, the Phaedrus web site has a wealth of technical information on the subject of compression and the operation of the Phamulus and its circuitry, which I recommend to anyone interested in the detail of these subjects.

In Use Once connected to the required +4dBu inputs and outputs, and having given

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it a good few minutes to warm up and settle down, operating the Phamulus is, as you’d expect from a variable-mu compressor, an intuitive experience. With an attack time of 50ms (as on the 436C) and a slightly shifted release time of 0.1s to 1.1s (as compared to the Altec’s range of 0.3s to 1.3s), the Phamulus’s overall performance is similar in character to that which I recall hearing from closer recreations of its original inspiration. As in other variable‑mu compressors, the compression ratio increases the more the incoming signal exceeds the threshold. So, with the attack time fixed, the end result depends on the interaction that you create between source, input level, threshold and release time. Like any good Altec 436-based compressor, the Phamulus excels in the indefinable art of pulling a source or a mix together sonically. However, the lack of a side-chain high-pass filter (or external side-chain input) means that I personally would only consider it for compression duties on bass-heavy tracks as a parallel processor, and in that scenario I did miss the presence of a wet/dry mix control. Where the Phamulus really worked for me was sitting after the preamp, compressing a vocal, acoustic guitar or bass as it was being recorded, giving me level control and a sense of relaxed cohesion — plus that bit of additional warm harmonic richness when I pushed the transformers. Used as an overall limiter on a mix stem it acquitted itself well but, without a side-chain high-pass filter, I doubt that I’d use it myself to master a track unless I particularly needed that old-school sound. Although I didn’t have a pair to play with, linking two

Alternatives Apart from two 500-series units (Retro Instruments’ Doublewide Tube Compressor and IGS Audio’s Tubecore 500), I couldn’t find a hardware vari-mu compressor at anywhere near the price of the Phamulus, so if you want that sound, the Phamulus is the value choice.

Phamuli (my Latin O-Level finally came in useful) in stereo is simply a matter of connecting them together using the supplied interconnect cable and setting the controls identically.

Conclusion To me, there’s something seductive about the sound of a good vari-mu compressor, and Phaedrus Audio’s Phamulus certainly has that attraction built-in. Although its circuitry is more than similar to that of the 436C on which it is based, its shape and size demonstrate not that is a clone of the original, but rather that Phaedrus Audio have used modern transformers, electronic components, materials and construction techniques to produce an Altec in a small box. Whilst the Phamulus doesn’t have the transformers to sound like a 436C or an EMI RS124, to my ears it does perform in a similar way and is, in its own right, a very good vari-mu compressor. Apart from a couple of 500-series products that aren’t directly comparable, the Phamulus is, by a long way, the most cost-effective example of that genre that I know; it offers a performance package that is seriously good value for money. $$ £599(about $850 when going to press). WW

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Studio Electronics Quadnic

Eurorack Digital Oscillator Module


he Studio Electronics Boomstar range has grown considerably of late, aided by a few cherry‑picked partnerships. Designed in collaboration with SpaceHardware, the Quadnic packs four digital oscillators into an impressively compact 12hp, its finger‑friendly design placing all the I/O beneath smoothly plush controls. No manual is supplied, so you’ll need to visit the Studio Electronics web site to examine all the possible configurations and to download the ‘quick guide’. In a nutshell, oscillators 2 to 4 can serve as slaves to the first or can act as masters that operate independently. In the latter case, there are individual CV inputs and audio outputs, plus a mix output too. However, given there’s but a single set of controls, independent operation can feel slightly awkward — as if you’re menu‑surfing but without the comfort of a display. From the outset you need to be

constantly aware of the bright blue LEDs denoting oscillator selection and master status. You also need to know about the startup values for each oscillator: specifically that only oscillator 1’s settings are read from the panel; the rest are zeroed at every power cycle. If preparing for a gig or if you simply like working on complex patches over several days, this behaviour is rather frustrating to say the least. Naturally, you can make any chord you like by manually tuning each oscillator, but for convenience, slaved oscillators can take advantage of a number of preset chords. The (notch‑free) Chord knob sweeps smoothly between major, minor, major sixth, minor seventh and

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dominant seventh, plus octaves and fifths to finish. Release any oscillator from slave status and the same knob becomes a detune, with an octave available to the first two oscillators, and two octaves for the second. There are four banks of 16 digital waveforms and since they too are selected by continuous controls, choosing by ear is the order of the day. Turning the tiny Bank knob you realise (for no descriptions or diagrams are provided) that bank 1 begins with pure waveforms before rapidly switching to thinner digital territory. Bank 2 is where the normal waves live — working through sawtooth and various pulses before heading straight back into in buzz‑land. Fans of PPG and Prophet VS tones will surely appreciate bank 3’s waveforms, which are also fascinating mod sources if transposed right down. This leaves the final bank, which appears to be a mixed bag but notable for its organ‑type and formant waves. Stacking different waveforms as chords or a monster unison is little short of remarkable, and in most cases, there’s usually something bizarre or malevolent lurking in the lowest ranges. For even more dirt and ‘oomph’, Drive acts as far more than a mere level control; it introduces hard clipping at any point beyond its midway setting. The Quadnic’s TARDIS‑like qualities were already becoming clear, but there’s more — supplied by the central Process Mode selector. This seven‑way control, tailored by its amount and bi‑polar CV pots, imposes wild and wonderful transformations whose effects vary from waveform to waveform. In the first process, a copy of the oscillator is

introduced and progressively detuned by the amount knob. The next also invokes a copy but one that performs amplitude modulation. Phase modulation by a triangle wave follows, resulting in a two‑operator flavour of FM; it’s followed by the same process modulated by a more complex wave, which can produce some searing sync impersonations depending on the bank and waveform being processed. Wave Sequencing scans the collection of 64 waveforms at a rate controlled by the Process amount. The final two are both phase distortion algorithms that sound approximately like mild and stronger sync in turn. While there’s just a single Process CV input, you can specify unique amounts of modulation and level for each oscillator. The greedy might wish that, when the individual oscillator CV inputs weren’t in use, they switched to being individual Process CVs. On the other hand, the Quadnic is probably mind‑bending enough already! This is a superb little module brimming with options and diverse tonality. It’s capable of Swarmatron‑like chord swarms or monster unisons and drones, but is equally happy when delivering four independent, complex voices. True, it employs smooth knobs in places where switches or buttons would have worked better, but for me the only real weakness is the ‘oscillator amnesia’ that occurs at each power down. What I wouldn’t have given for a single ‘patch memory’! However, in all other respects, the Quadnic is hard to fault and comes highly recommended. Paul Nagle $$ $269 WW

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AJH Synth MiniMod Transistor Ladder Filter Eurorack Filter Module


or many, an analogue synthesizer is defined by its low‑pass filter. There have been numerous notable examples of this, including the ARP4023 filter that separated the ‘white face’ Odysseys from their siblings, the Korg35 filter in the early MS20s, the SSM2040 that elevated the first Prophets above their successors, and the ‘matched‑pair’ ladder filter that helps the earliest Minimoogs to command such remarkable second‑hand prices. Of these, perhaps the most revered (and certainly the most emulated) is the Minimoog filter, and it’s the earliest version of this that forms the basis of the Transistor Ladder Filter in the MiniMod. Like the MiniMod VCO reviewed in the June issue of SOS, the VCF takes the original specification and extends it in interesting ways. But before the company could implement any improvements, they had to get the underlying response right, and that proved trickier than expected. Numerous prototypes were tested before it was discovered that it was necessary to reduce the power supply voltages and signal levels within the filter core to match the Minimoog’s but, once the underlying architecture was in place, it was then possible to add the extra facilities without changing the filter’s character. To recap, the Minimoog’s filter section offered cutoff frequency and emphasis controls, four key‑tracking amounts, an ADS(D) contour with an Amount control, and a single modulation input pre‑patched to the synth’s modulation section. The AJH goes much further. First, there’s

a three‑channel audio mixer that accepts signals up to 16V peak‑to‑peak before clipping, which means that you’re not going to have problems injecting signals from elsewhere. Following this, there’s the filter itself. This offers frequency and emphasis knobs, of course, but the cutoff frequency can also be modified by signals presented to four CV inputs: a variable input with a maximum response of around 0.75V/oct, a 1V/oct input, a 1/3 input, and a 2/3 input, the latter two of which can emulate the Minimoog’s keyboard tracking switches. (There’s an error in the manual, which states that these respond at rates of 333mV/oct and 666mV/oct respectively, but that should read 3V/oct and 1.5V/oct.) There’s also voltage control of emphasis, which is always welcome.

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Finally, there’s an internal jumper that, when removed, roughly doubles the level of the signal within the filter core for a more overdriven sound. Before passing any audio signals through the VCF, I put the MiniMod next to my Minimoog and tested whether their two filters self‑oscillated in the same fashion. I was surprised to find that the self‑oscillation of the AJH extended to much lower frequencies than that of the Minimoog and, since the reduction of emphasis at low frequencies is an important factor in the Minimoog’s bass sounds, I took some measurements and sent them to AJH Synth. It seems that a faulty capacitor was to blame and, within hours, a replacement was winging its way to me. Inserting this into the MiniMod rack, I repeated the tests and found that all was as it should be. Next, I took the outputs from the three MiniMod VCOs and directed these to both the AJH VCF and the Ext input on the Minimoog. Happily, I was always able to make the two filters respond almost identically. I could also make them sound different (for example, the self‑oscillation on the AJH is ‘stronger’ at the highest emphasis values than that of my Minimoog) but that’s not the point — the AJH could sound like the Moog if I wanted it to. I then pushed it beyond the Minimoog’s capabilities. Using an Analogue Systems Sorceror as the source of the numerous CVs, Gates and modulators that I needed, I directed one of the MiniMod’s contour generators to the VCF’s variable input, four 1V/oct outputs from the AS keyboard to the filter’s 1V/oct input as well as to each of the three MiniMod oscillators, the

second MiniMod contour to the filter’s 2/3 input, an AS LFO to its 1/3 input for some subtle modulation effects and, finally, an AS contour generator to the Emphasis CV input for some less subtle modulation effects. (Phew!) Despite having nothing more than the MiniMod’s own VCOs, VCF and VCA in the signal path, the range of sounds that I could obtain was superb, ranging from Minimoog leads and basses to classic Moog Modular patches to even more extreme sounds and effects. Oh yes... and passing the sounds through a spring reverb was a revelation. With a little vibrato and portamento, it was 1972 all over again. With so much on its panel, the MiniMod VCF is sometimes a bit fiddly to use, but it retains the company’s standard layout with all of the sockets beneath the knobs, which is sensible. Like the MiniMod VCO, it’s available in black and silver finishes and, as before, I would recommend the black ‘un, which looks a whole lot smarter. It’s not the cheapest Minimoog‑inspired filter you can buy, but it’s far from the most expensive. I like its look and feel, as well as the service provided by the manufacturer. And, of course, it can sound just like a Moog filter, whether used to create euphonic sounds or terrorised shrieks. What’s not to like? Gordon Reid $$ $299 WW

Mutable Instruments Rings Eurorack Resonance Module


utable Instruments’ Rings introduces the idea of virtual strings to generate its own flavour of polyphony. Familiar from the


outset, Rings closely resembles the resonator section of the Elements module. It supports three resonators: physical modelling processes that take unpitched sounds such as clicks and plucks and generate harmonic, musical tones in response. Well, mostly. Sometimes the resonances created are metallic and inharmonic, just the job for percussion, tuned or otherwise. Rings does strings — and so much more! Two buttons toggle the module’s polyphony and choice of resonator. For the former, you have a choice of one, two and four simultaneous notes, but you won’t see separate CV inputs for each voice, nor audio outputs. Polyphony in this sense means that several notes are heard at once, their tones decaying together like the strings of a guitar or harp. The trigger input, which responds to voltage or audio pulses, is even named ‘Strum’ to reinforce the idea. The duration of ringing, which is convincingly non‑synthetic at times, is set by the Damping control over a range 100ms to10s. The top right‑hand button toggles between the resonator states, each indicated by a colour: green, amber or red. Vital here are the Structure and Position knobs, with each behaving differently according to the selected resonator. Spinning the Structure knob is a smooth and satisfying experience thanks to a slew process that wards off the otherwise rough and ugly transitions. Similarly, some of the CV inputs also have a slew limiter, preventing clicks and removing A-D conversion noise which would otherwise rough up the sensitive internal parameters. Of the other controls, Brightness sets the level of upper

harmonics, turning simulations of glass and steel into wood and nylon as you turn anti‑clockwise. First up is the Modal Resonator. Similar to the resonator of the Elements module, it mimics the resonances of various vibrating structures — strings, plates, etc. The Structure control sets the frequency ratio of generated partials and delivers an assortment of plucked strings until approximately its mid‑point, where the output becomes more bell‑like. Chimes, electric pianos and xylophone‑ish creations are all easily achieved too. By tweaking the Position — the point on the virtual surface at which the exciter is struck — the output becomes more nuanced, giving a tangible sense of an object being hit in different ways. Sympathetic Strings follows, modelling the resonances that occur not in strings that are struck directly, but in those nearby. The Structure knob sets the intervals between the sympathetic

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strings. Natural‑sounding phasey sweeps appear when you feed random voltages into the Position’s CV input. Supply an LFO and you get rich chorus. This can be the most subtle of the resonators, producing sonic images of not‑yet‑invented string instruments as well as acoustic guitars, kotos and sitars, with a smattering of FM‑style jangling harpsichords and clavinets en route. Adding randomness to the Structure CV input is not for the faint‑hearted though, the string detune effects instantly erasing my claim of subtlety. The final type, Non‑Linear Strings, is Karplus‑Strong‑based and loaded with unruly inharmonics. It’s probably my personal favourite and covers territory ranging from eerie PPGs and impressions of oscillator sync to tuned percussion, hang drums, ringing monochords, gamelans and lots more. Regardless of the selected resonator, stimulation — in the form of an exciter signal — is required to produce musical tones. Ideally, it should be introduced via the audio input, and I got plenty of mileage from analogue percussion, synth blips, white‑noise bursts and even vocal samples. Pretty much anything is worth a shot, and if you don’t supply an input, Rings will create a suitable signal for you automatically. This might seem like a limitation if you’re used to the triple exciter engine of the Elements*, but I found it liberating. Even working with only the default internal exciters, the potential is enormous. Signals — audio or voltage triggers — sent to the Strum input initiate new notes. They don’t need to have any correlation with the (1V/oct) note voltages received at

the pitch input. In fact, you don’t actually need triggers at all; Rings will helpfully generate a note for every new pitch it detects at the input. However, if you do dispense with them, you’ll notice that repeatedly playing the same note won’t generate any new triggers. The output stage offers a useful stereo split. It’s divided into odd and even outputs that behave differently depending on whether Rings is in mono or poly mode. When mono, the outputs carry odd and even numbered partials from the Modal Resonator, or the de‑phased components if either of the others is selected. In poly mode, the outputs switch to the odd and even numbered strings or plates, which is lush, trippy and an easy justification for that extra mixer channel. At approximately half the price of the Elements, the compact (14hp) Rings is way more than just its ‘resonator bit’. Polyphony brings qualities not attainable with the older module — qualities well suited to the type of instruments being synthesized (there aren’t many single‑string guitars after all!). Stripped down to a trio of resonators, Rings might have felt underwhelming after the more complex Elements. Instead it’s something of a bargain, with almost every combination of its settings producing gold. Paul Nagle

*The Elements module has the same CPU power available, but spends it not on polyphony but on reverb, multiple exciters and a slightly more complex resonator. Rings’ extra resonators were added to it in firmware version 1.1. $$ $359 WW www.mutable‑

YOUR GEAR. YOUR STORY. From classic keyboards to modern mini synths, plus the vintage and boutique effects that take them to the next level - has the gear you need to bring your sound and your story to life.



Triad‑Orbit Mic‑stand System This beautifully engineered system reaches the parts other mic stands can’t...

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Orbit System



egular SOS readers will know that I get disturbingly passionate about the engineering of one of the most mundane studio peripherals: the microphone stand. But I’m sure everyone has experienced the frustration of drooping mic stands, or stands that wobble or topple over at the slightest provocation. Sometimes these problems are caused by ‘operator‑error’, setting up the stand incorrectly or misusing it, but often it’s just because some mic stands are very poor designs and can never work well or reliably over long timescales. Inevitably, a well‑designed and well‑made studio stand — one which will genuinely last a lifetime — is never going to be ‘cheap’. But if the cost is amortised over 20 years or more it will probably work out less expensive than replacing broken stands every year or two, and will give you far greater peace of mind knowing your mics are supported securely and reliably! For the manufacturer, a benefit of a higher‑cost stand is that the engineering can be much more developed and far better executed. This is certainly evident in the design of the Triad‑Orbit mic stand system — and it really is a ‘system’, with all manner of compatible elements making this probably the most versatile mic stand system I’ve seen to date. Every aspect has been developed and optimised in America by a group of professional musicians and experienced audio engineers to deliver a ‘no‑compromise’ modular mic‑stand system that’s both incredibly versatile and extremely robust, but still easy to use — exactly what every mic‑stand user wants!

Every element in the Triad‑Orbit system is compatible with every other element, and stands can be constructed in a mix‑and‑match manner to meet any specific requirements. The core of the Triad‑Orbit world comprises four sizes of basic vertical mic stand. The TM (mini), T1 and T2 all have one extendable section, while the range‑topping T3 and T3C have two extendible sections. The ‘C’ suffix indicates the attachment of (removable) three‑inch locking castor wheels, to create a rolling studio stand. There’s something available to cover every normal studio requirement, and I’m sure more will be added in time. So rather

Triad-Orbit PROS

• The Quick‑Change component coupling system is genius! • Very sturdy clutches and clamps, providing very secure positioning. • Unusual but versatile multi‑angle folding leg facility. • Bizarre but extremely useful dual‑arm boom options. • Neat ball‑swivel mic ‘hanger’ accessories. CONS

• No option to change boom counter‑balance weights. • High cost. SUMMARY

An extraordinarily versatile and comprehensive mic-stand system, with a wide and unusual range of accessories and attachments, all designed to last a lifetime.

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than list myriad mind‑boggling dimensions here, I’ll leave you to check the company’s web site for the minimum and maximum heights, as well as the weights and folded sizes of each stand variant. It should be noted that these stands are all unusually heavy for their sizes, and that most of the weight is kept low in the base, to maintain a very stable centre of gravity. It takes a considerable effort to topple one of these stands! All of the core mic stands have a familiar‑looking three‑legged base arrangement, and the legs are replaceable should they become damaged (which seems unlikely). The folding tension can also be adjusted using a couple of supplied Allen keys (or ‘hex wrenches’ if your English is of the US variety!). Naturally, the legs can be folded down completely for storage, but they can also be positioned and locked individually via a foot‑operated latch at any of four intermediate angles up to 65 degrees. If all three legs are moved to one of these midway positions the stand’s footprint is usefully reduced with only

a small reduction in stability, while the base clearance and overall stand height are both increased. Again, check the Triad‑Orbit web site if you want full details, but hopefully I’ve been able to give you a feel for how this all works. The ability to alter the leg angles to intermediate positions helps with interlacing stands among each other (or for storage), but also makes it easy to cope with uneven surfaces (eg. steps or low platforms). Unusually, it also allows the stand to lean in one direction, which proved surprisingly useful in gaining easy access to instruments or drums without having to use a boom arm — something I found particularly useful. Moving up to the clutch grips for the extendable section(s), these are again of a fairly traditional design using a simple split‑collet under the metal screw barrel, which is covered with a high‑grip knurled rubber sleeve. The operation is familiar and easy, and felt strong and reliable. Another clutch grip is attached to the base section, allowing the legs to be slid

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along the lower tube to reduce the overall length for storage. Perhaps the most attractive engineering feature of the Triad‑Orbit design is the integration of a bespoke ‘Quick‑Change’ (Q‑C) coupler at the top of every stand. This comprises a hexagonal socket which accepts a short mating bar attached to a standard 5/8‑inch screw thread. (A chunky metal 3/8‑inch European thread adaptor is also included). The release mechanism is operated simply by pulling a knurled metal sleeve away from the top of the fitting, releasing the pressure applied by a steel ball‑bearing inside. A new Q‑C attachment can be pushed directly into an empty socket whereupon it locks instantly

in place, as any effort to pull the spigot out causes the ball‑bearing to cam tighter into the spigot. This is a very elegant and effective system that makes the Triad‑Orbit system a joy to use and configure, allowing very rapid fitting and releasing of mics already in their mounts, or to allow other mounting accessories to be changed in seconds, including a variety of boom arms and other fittings.

Boom Arms Most Triad‑Orbit users will require a boom arm on their stand, and there are two standard options, Orbit Mini and Orbit 1. Both have two clutch mechanisms, one controlling the extending part of the

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arm and the other the position of the arm within the stand mount (to vary the balance and/or reach). Again, the business end of the boom arm features another Q‑C mechanism, and the stand mount is attached to the stand via a Q‑C spigot. The two boom models only differ in size, with the Orbit Mini adjustable between 310 and 550 mm (12.2 to 21.65 inches), and the Orbit 1 spanning 530 to 950 mm (20.8 to 37.5 inches).

“It’s a great system for positioning mics around a drum kit... or any situation where you need to support multiple mics without cluttering up the floor space.” Setting the Orbit boom apart from its rivals is its use of a substantial stainless‑steel ball‑swivel mount attaching the boom arm to the stand mount, in an arrangement similar to many camera tripods. A large T‑bar handle adjusts the tension and clamps the arm solidly in place once positioned. This configuration works very well indeed, and has the rather curious property of allowing the boom arm to be positioned off to the side of the mount, rather than always directly above it, which might be useful on occasions! Again, nothing here is lightweight, with the Orbit Mini weighing 1.93kg (4.25lbs) and the Orbit 1 a chunky 2.15kg (4.75lbs). Among all this lovely engineering one trick does seem to have been missed, though, which is the absence of alternative counter‑balance weights. The balance

weight on each boom is secured with a grub screw, so could be user‑replaceable, but as yet alternative balance weights aren’t available as accessories. I’d like to see this option introduced to allow easier counter‑balancing of different mics and attachments — as some other manufacturers already do.

Into Orbit At this point the Triad‑Orbit system diverges from the relatively conventional mic-stand paradigm and into a world of the weird and wonderful, as perfectly illustrated by the Orbit 2 ‘dual‑arm’ boom. As the name suggests, this comprises two short boom arms, each with an extending section, and each attached via ball‑swivel joints to the triangular stand hub (shown in this review’s main picture). This hub has a centre‑pivoting action to allow an additional 72.5 degrees of tilt left or right, and attaches to the stand with the now‑familiar Q‑C spigot. Each arm can be extended between 420 and 660 mm (16.5‑26 inches) from the ball swivel and, taking the width of the centre hub into account, that affords a maximum horizontal span between the two arm tips of 1487mm (58.5 inches). The Orbit 2 weighs 2.15kg (4.75lbs). There’s also an Orbit 2x version, which is very similar but the arms are replaced with Q‑C sockets directly mounted on the hub ball‑swivel joints, and it is supplied with a pair of plug‑in short arms (242‑320mm/9.5‑12.6 inches) and a pair of long arms (395‑632mm/15.5‑24.8 inches). Inevitably, while this arrangement greatly increases versatility, the extra joints also reduce the maximum load capability slightly, but the Triad‑Orbit web site is quite helpful in providing the

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relevant shear and loading weight limits, and it’s more than strong enough for most typical requirements. Although they look rather ungainly at first, the Orbit 2 and 2x boom arms are incredibly adaptable, with countless applications only really limited by the imagination. It’s a great system for positioning mics around a drum kit, for instance, or any situation where you need to support multiple mics without cluttering up the floor space. As I hinted before, the Triad‑Orbit system also features a number of accessories and adaptors which further expand system’s adaptability. For example, the OA adapter is a stand‑alone swivel joint with a Q‑C spigot at the base and a Q‑C socket extending from the ball‑swivel, allowing mics to be positioned at any desired angle very easily. Then there are the M1 and M2 Micro Orbital adaptors, which are intended as ‘hangers’ to suspend a mic below an overhead boom arm. These two

models comprise a smaller version of the ball‑swivel, and only differ in the length of the extending arm, which terminates with a 5/8‑inch thread. The longer version gives greater clearance for large‑diaphragm mics mounted in cat’s‑cradle shockmounts. There is also a variety of alternatives to conventional floor stands, such as a screw‑down desktop base and a pipe‑clamp, both with Q‑C sockets to accept any of the boom arms and other adapters. A comprehensive collection of different Q‑C spigot mounts are also available as alternatives to the standard 5/8‑inch threaded adapter supplied with the stands and boom arms. For example, a heavy‑duty version has a notched spigot which allows heavier loads to be supported, and there are two versions with 1/4‑inch threads for camera equipment, one with an integrated lighting equipment stud. If the Quick‑Change release mechanism appeals but you can’t justify changing

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your entire mic stand collection, Triad‑Orbit have a solution for that too, with the ‘Retro’ components. The IO‑R, for example, screws directly onto a conventional mic stand or boom arm (and is locked in place with a grub screw) to bestow it with a Q‑C socket, while the M1‑R and M2‑R do the same but with ball swivels and drop arms. If you need to mount an iPad or iPhone on the stand, there are adaptors for those as well (and the Orbit 2 dual‑arm boom makes a great mounting system for a mic and iPad!). When increased floor‑stand stability is required, the optional Triad‑Orbit GB‑3 ‘Grav Bags’ can be filled with lead shot and attached with velcro tabs to the stand feet. And if additional height is required, the T‑ES Elevator Shaft can be attached to any of the base stands to provide an extra 838mm (33‑inch) of height, with Q‑C fittings at each. When fitted to a T3 stand this makes a substantial and effective ‘Cathedral Stand’, raising mics above orchestras and choirs, or for organ lofts and the like. Finally, two variations on padded carry‑bags are available to protect and transport complete stand systems, the deluxe version having roller wheels which make it a lot easier to transport a heavy stand system!

Impressions I found the Triad Orbit system a joy to use. The core components are strong and robust, and the clutches and clamps all provide very secure and stable positioning. As I mentioned earlier, I’d have liked an option to swap out the boom counter‑balance weight — most stand ‘drooping’ and instability issues are caused by not balancing the boom arm

Alternatives The Latch Lake MicKing stand system is equally sturdy and well‑designed, while the K&M top‑line systems are well‑known as reliable and easy to maintain. I also like the Sontronics Matrix, which is an adapted lighting stand, and has the benefit of being relatively lightweight but extremely strong.

correctly — but that issue aside the basic stands do a superb job and I’m sure would provide a long service life. The icing on the cake is undoubtedly the elegant and highly effective Quick‑Change mount, which makes attaching and removing microphones a very safe and easy process; the ability to add Q‑C mounts to legacy mic stands is a very attractive one. The rather wacky dual‑boom arm proved its worth very quickly when rigging mics around a small drum kit, and I’m sure this unique facility will serve to persuade many potential customers of the many benefits of the Triad‑Orbit system all on its own! There’s no getting away from the very high cost of this system, sadly, but for those that are prepared to contemplate such expenditure these stands are unlikely ever to need replacing, and are amongst the most robust and secure I’ve ever used. The Triad‑Orbit system is also one of the most versatile and comprehensive mic stand systems I’ve ever used. I am very impressed with the overall engineering design, and these stands afford great peace of mind that microphones are supported safely and securely. That’s worth a lot in my book! $$ T1 Short tripod stand $139, T2 Standard

tripod stand $159.99, T3 Tall tripod stand $199, T3C (tall with castors) $219. Check with dealers for details of accessories. WW www.triad‑

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Highly Recommended Clarett has been widely acclaimed since its release, with great reviews, with accolades and awards all round from the likes of Sound On Sound, Resolution and Studio One Expert.

...all you’d want in a high-end conversion box, but at about half the price.

... a smooth, warmed texture to vocals when Air is engaged. Brian Kennedy (Producer/Songwriter credits include Brandy, Chris Brown, Jesse McCartney, Rihanna, Jessica Mauboy, Natasha Bedingfield, BoA, Cheryl Cole, Backstreet Boys, Jamie Foxx, Nelly, and Westlife.)

Glenn Rosenstein (Producer, Mixer, Engineer, Songwriter- credits include U2, Madonna, Talking Heads, The Ramones, Lisa Lisa & Cult Jam and many others.)

...simplified my home studio remarkably Brandon Boyd (Incubus)


Behringer X‑Touch & X‑Touch Compact Control Surfaces Behringer’s X‑Touches offer affordable DAW control and tight integration with the company’s X‑Air system. SIMON SHERBOURNE


ehringer’s X‑Touch controllers were first previewed in 2014 and eventually started shipping towards the end of last year. Since then they’ve had a significant update that expands on their DAW control functionality, adding direct integration with Behringer’s X‑Air series live systems. I tested the full‑size X‑Touch and the X‑Touch Compact. The main model

is essentially an emulation of the Mackie Control Universal, plus the X‑Air control, with Ethernet, USB and MIDI connectivity. The Compact is a more generic MIDI controller, although it also features a subset of the Mackie protocol and can extend a full X‑Touch. There’s also a budget, portable ‘Mini’ device that shares the X‑Touch name. Physically, the X‑Touches are best characterised as ‘chunky’. While they actually have a modest footprint, they are

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rather thick. This puts the faders a little high for comfort when the unit is on the desktop, and also means that the scribble displays are at an acute viewing angle and difficult to read from a seated position. I ended up raking the X‑Touches to a much more usable angle using that studio essential: the Ikea laptop stand. Size‑wise, the Compact and full‑sized models differ only in width, allowing them to sit side‑by‑side and form a larger control surface. Connectivity options are generous: there’s a USB connection that will suit most DAW control scenarios, and both X‑Touches have a built‑in hub for connecting a further two USB devices. Both models have full‑sized MIDI In and Out ports, and multiple quarter‑inch footswitch inputs (two on the Compact, three on the main). The bigger brother goes an extra step with an Ethernet port which, as well as providing connection to an X‑Air, allows connection to a computer via MIDI RTP instead of USB if required.

Connecting DAWs I tested the X‑Touches with Logic, Pro Tools, Live and Reason. Of these, Logic enjoys the deepest level of functionality with the controllers due to its extensive support of the MCU protocol. Configuring the X‑Touches with Logic, Live or Reason amounts to telling them that Mackie Control devices are connected, and booting the surfaces in ‘MC’ mode. Pro Tools control is via the more limited HUI protocol, which is only available on the larger X‑Touch. Once configured, the X‑Touch works with Logic pretty much exactly like an MCU. At this price, you might not expect wonders from the nine touch‑sensitive, motorised faders, but these aren’t bad, with a reasonably smooth action. When playing

Behringer X‑Touch & X‑Touch Compact $599/$399 PROS

• Touch‑sensitive encoders and motorised faders. • Affordable. • Powerful combo with Behringer X‑Air series mixers. • Network connectivity. CONS

• Not exactly slim. • No Mac editor for the X‑Touch Compact. • No MIDI mode on the larger X‑Touch. • Limited HUI (and therefore Pro Tools) support. SUMMARY

The X‑Touch is an expandable and affordable MCU clone, with network connectivity and powerful integration with Behringer digital live systems. The Compact offers generic USB MIDI control as well as a decent chunk of MC functionality, but is crying out for a Mac editor.

back automation they’re a bit clattery and juddery, and I occasionally had some buzz from the actuators, but this seemed to improve with the latest firmware. Most importantly, they are snappy to update when banking or nudging across the DAW mixer. The jog wheel has a cool illuminated ring, and does the job, although it has some wobble and is one of the more light and plasticky feeling parts of the surface. The transport buttons are solid, however, and all the other buttons (which are rubber) feel pretty good too. The buttons go one better than the MCU by being internally

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illuminated to show their status. The other tweak compared to the MCU is better metering, with each channel strip sporting an eight‑segment meter. This improves on the single Signal Present light on the MCU, although is lower resolution than the meter bridge on the original Mackie HUI.

Logic & Cubase X‑Touch functionality with most DAWs varies mainly in the way each program implements MCU. Logic and Cubase both have full support, with similar features and workflows. In these hosts, the labels on the X‑Touch’s buttons generally do what they say, which is not always the case in other programs. (The more generic Compact is mostly unlabelled — for other differences, see the ‘Custom Compact’ box). Above each of the main eight faders are the obligatory Select, Mute, Solo and Rec

buttons, then there’s a small display and a rotary encoder. The displays generally show track name in the top half and a rotary value in the bottom, unless the rotaries are all focused on a single track. In the promotional pictures of the X‑Touch, the displays are all lit up with different colours, but in all my DAW testing they were black and white. Eventually I figured out that the colours are only implemented in X‑Air control mode and colour coding is not supported via MCU or HUI modes. The encoders default to pan, but can be re‑assigned to EQ, sends, plug‑ins or instruments with the row of mode buttons at the top. The first press of any of these will spill a single parameter of the chosen type to each track; a second press will switch the pots into a channel focus mode, where all the pots control a single object. For example, pressing EQ once brings up low-

Custom Compact Physically the Compact has a lot in common with the daddy X‑Touch, allowing it to be used as a side‑car to expand in groups of eight. However, while the main X‑Touch is clearly conceived as a MCU clone and X‑Air controller, the Compact is a generic USB MIDI controller that has a good MCU mode. In place of the extensive transport, jog wheel and button array on the right of the X‑Touch, the Compact has basic transport, ‘layer’ buttons and eight additional rotary encoders. In MC mode, six of the encoders can be pushed to select the mode of the main track encoders: pan, EQ, send, etc. The last two knobs serve as bank controls: turning these encoders scrolls the faders across your DAW mixer. This is a novel approach that works more smoothly in

some hosts than others, and feels a little less precise than bank buttons. The six mode select knobs serve no other function in Mackie mode, which seems a waste of good encoders. I wished I could use these knobs for channel focused functions like Logic EQ, or Live Device Macros while leaving the strip encoders on pans or sends. Here, I guess the device is working within the limits of the MCU protocol. The big difference with the Compact X‑Touch, though, is that it can switch out of MC operation into a user‑definable MIDI mode. In this mode, the Compact offers two layers of MIDI Note and CC assignments. These can be edited with a downloadable app, although Mac users are out of luck: it’s Windows only.

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cut frequency for the Channel EQ on each track. The cursor buttons can then be used to step through different parameters and bands. Tapping EQ again will redirect all the pots to the EQ on the selected track. Sends are a little awkward in that the default first assignment is the routing destination of send 1, instead of the level. A right cursor press is required to get to this. The up/down cursors step you through the available send slots. A Flip mode is available to bring the rotary values down to the faders. The rotaries are also push buttons, allowing them to be used for confirming routing assignments in Send mode and selecting inserts in Plug‑in mode. In other contexts, pushing a rotary resets its assigned parameter to the default value. The main display shows timeline position and the current encoder assignment page. Below this is a row of View buttons that show different track types in the mixer. The other most significant key button cluster is the Automation section, allowing you to set the selected track’s automation mode. You can also hold a mode button and select multiple tracks, or hold the Option button to change all tracks.

Universal Control? Ableton Live functionality is again derived directly from Mackie Control, simply by choosing the preset in Live’s Control Surface settings. As well as the usual channel strip and transport functionality, you get knob modes for controlling pans, sends and plug‑ins. For the latter two there’s only a selected track focus view, but I didn’t really miss the one‑parameter‑per‑track mode. Plug‑in mode gives you fairly easy access to Devices, with the default assignment on Racks going to the eight Macros. If anything, this works more smoothly than in Logic, although bringing up a device on the surface doesn’t automatically show it on the screen. However, the Automation button cluster is re‑purposed for views, one of which is Device/Clip view toggle. Predictably, Propellerhead Reason has a different take on Mackie Control implementation. In Reason each device has its own map, and the controller ‘points’ at whichever device has MIDI focus. As the up/ down cursor buttons are mapped to track selection in the Sequencer it’s pretty easy to move around. You also have the option

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to lock the X‑Touch to any device, including the main mixer. The buttons and knobs do completely different things for each device, but the mappings are sensible. The ReDrum set is particularly fun, with channel select buttons triggering each sound, and the Rec buttons initiating sample recording directly into the drum channels. In the main mixer, the mode buttons move the knobs around to the different sections, and the channel select buttons put the rotaries into single track focus. If you’re a Pro Tools user, you’ll already know that the MCU protocol is not supported, but HUI is, and this is available as a mode on the larger X‑Touch. The functionality is somewhat sparse, unfortunately. If basic fader and transport

control is all you need then all is well. Unfortunately the more intelligent rotary mappings are not implemented, despite these being supported by HUI. Instead the mode buttons access control of pans and the first three sends; there’s no plug‑in control whatsoever. You do get control of track automation modes, and the View buttons are used for Auto Enables, but that’s your lot.

Out Of The Box Live-sound production has been rapidly transforming of late, with a move to compact digital back‑ends controlled by tablets and network‑attached control surfaces. Behringer have been early to jump into this market with their X‑Air series

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products. These devices look like stage boxes, but feature mix and effects engines, all controlled from a tablet or laptop. The X‑Touch can now be connected to an X‑Air rig to add more traditional physical fader and knob control. Connection can be direct via an Ethernet cable, or via a switch that’s connected to the X‑Air. You can direct‑connect an X‑Touch and also connect an iPad/Android tablet via the X‑Air’s on‑board WiFi, which gives you multiple control points, and also provides more visual feedback than the X‑Touch alone. The implementation is deep. As well as bankable fader control, you can dig in and edit EQ, dynamics, cue and effects sends on any selected channel using the top mode buttons. The four Modify buttons and six Automation buttons are used for spilling the four effects sends and six monitor sends to the faders, providing a quick way to manage monitor mixes. One thing that would really make it fly would be if iPad and/or laptop software displays could follow the mode selected on the X‑Touch, but although these all stay in sync they are essentially independent. A real plus is the ability to control a DAW and an X‑series device at the same time: one of the reasons you might use a network switch rather than connecting directly. In this configuration, the X‑Touch’s transport stays locked to your DAW, and you can toggle the rest of the surface between X‑Air and DAW control at the touch of a button. Obviously this would be really handy in live situations with backing track playback, but could also be useful for recording.

Alternatives The full‑size X‑Touch is unique in offering MCU‑powered DAW control alongside X‑Air integration, but if it’s just the DAW side you’re interested in then there are some alternatives to check out. The second generation of Icon’s QCon Pro surface, the Pro X, is due to be shipping by the time you read this. This has a similar spec to the X‑Touch, with an attractive low profile and a full meter bridge (the latter could be a pro or con depending on your preference). Based on some pre‑order listings it will cost around $100 more than the X‑Touch. Mackie’s own Control Universal Pro is of course also still available. If you need something more customisable, with fader recall groups and Pro Tools support then look at Avid’s Artist controllers. On a smaller budget, if you just need a MIDI controller with motorised faders and can live without MCU mode, Behringer have the BCF2000 for around $300.

particularly revolutionary as far as DAW control goes, but it does have an aggressive price tag and some nice touches like the meters and illuminated buttons. If you want a motorised mix controller for Logic or Cubase then you’ll likely be happy with the X‑Touch. Functionality in Live and Reason is also quite extensive; Pro Tools users will be less impressed, although it’s still useful. If you’d rather have something that’s more customisable, the Compact, which can switch between MC and MIDI modes, may be a better fit, although the lack of a Mac editor app is an issue. The excellent integration with the X‑series live products is a stand‑out feature, and makes a tempting package for gigging musicians with a home studio.

Conclusion As the X‑Touch uses the Mackie Control protocol it doesn’t offer anything

$$ X‑Touch $599, X‑Touch Compact $399. WW

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PLUG‑IN FOLDER Polyverse Music I Wish Formats: Mac & PC VST, AAX & RTAS; Mac AU

I Wish is a granular synthesis plug‑in that uses incoming MIDI notes to play very short loops of audio. It was developed by Polyverse Music in conjunction with psytrance duo Infected Mushroom; the rather odd name is taken from one of their tracks. It was made available at a discount during the beta phase with a promise of free future upgrades. Beta or not, I had no difficulties under Logic (10.0.7) and once I’d checked the Quick Start section of the manual, I never looked back. To get that first sonic hit, you create an I Wish instrument track, then open the plug‑in and select an audio track as the side‑chain input (I began with a drum loop). Then it’s just a matter of starting your DAW and playing notes into the plug‑in to see what happens. It’s probably a good idea to

mute the source audio so the processes are audible in isolation. There’s a simple amp envelope, plus a wet/dry slider to set the balance of incoming signal and resulting synthesis. There are two large controls: Pitch and Formant. Pitch offers two octaves of shift up or down, and works in combination with the MIDI notes, which dynamically modify the loop length. Formant adjusts the audio playback speed and shifts the overtones, also in a range of two octaves either way. With just these controls and the notes you play, I Wish is an innovative tool for robotising, re‑pitching and granular‑freezing audio, and it becomes even more flexible when you assign a MIDI controller to the Formant parameter. Pitch and Formant can be linked together for producing drops and tape‑like effects. This is such a straightforward plug‑in that you might not even notice the dozen or so presets. They’re worth July 2016 / w w w . s o u n d o n s o u n d . c o m

a look, though: I chose one called ‘for drums’ and my loop became a freaky, pitched groove that screamed like a wounded android when I played high notes and stuttered like a mucky CD when I played low. But it was when I swapped my drum loop for vocals that I Wish really came to life. Sticking with low pitches, my puny voice sample became a rich source of robotic, glitching bass lines, like Tomita on acid! Naturally,


the results are very source‑dependent, but at times I Wish can sound like a vocoder or wavetable synthesizer, with up to 32 notes of polyphony if your system can handle it. There aren’t many other options to get sidetracked by. The X‑Fade control is useful for smoothing the loops generated, and there’s an LFO with the usual waveforms which can sync to the host clock. Along with an envelope this can modulate the pitch or formant in a bipolar fashion. Far more practical than a penny tossed into a fountain, I Wish will grant your desires assuming these involve replicating some of the godlike production feats of bands such as Infected Mushroom and Shpongle. Sometimes it reminded me of the Roland V‑Synth or Audio Damage’s Replicant, but always with its own distinct flavour. The trippy, artificial tones and glitches will be a gift to any psytrance devotee but they’re applicable to any genre where wilful experimentation is rife. Paul Nagle $$ $99. WW

Waves Abbey Road Reverb Plates Formats: Mac OS & Windows VST, RTAS & AAX; Mac AU

Plate reverbs are notoriously difficult to model because of their very fast ‘build up’ time and because of the complexity, density and subtle coloration of their reverb tails. Spare a thought, then, for the software engineers at Waves who took on the challenge of modelling four modified EMT 140 plates used at Abbey Road Studios, which can be heard on albums from countless top‑shelf artists, from the Beatles to Pink Floyd. EMI’s Central Research Laboratories designed hybrid solid‑state pickups and amps for plates A, B and C to lower the noise floor, while Plate D has all‑valve circuitry with a rather different character. In each case, Waves have modelled the harmonic distortion of the driver, pickups and output amps as well as the acoustic properties of each plate, including the way it interacts with the dampers used to reduce or increase the reverb time. In an July 2016 / w w w . s o u n d o n s o u n d . c o m

actual EMT 140 plate, which measures around 8 x 4 feet, a thin steel plate is suspended vertically on springs allowing it to vibrate when fed from a magnetic driver mounted near the plate’s centre. Acoustic waves fan out from the transducer and then reflect back from the plate edges, quickly producing a very dense reverb with a slightly metallic coloration. Two transducers configured as pickups are mounted between the excitation driver and the plate edges to produce a pseudo‑stereo output, which is then fed to a pair of output amplifiers. Reducing the reverb time is achieved using a fibreglass damping panel hung parallel to the plate but not touching it. Reducing the gap shortens the reverb time while moving it away increases it. All this has to go in a sound‑isolating box, making it an extremely bulky and heavy piece of studio kit. The Waves Abbey Road Reverb Plates plug‑in GUI provides a pictorial representation of a plate, with some very retro‑looking controls arrayed below it. When active, ‘vibrations’ can be seen emanating from the


driver and moving over the plate, but clicking it converts these to a waveform display. The plug‑in can be instantiated in mono, stereo, or mono‑in/stereo‑out versions. Old‑school quadrant faders set the input and output levels, though the bar‑graph level meters are bang up to date. A rotary wheel that looks like something from a submarine hatch selects which of the four plates is active; apparently this graphic was inspired by the original Plate Damper wheel. Plus and minus buttons adjust the virtual damper position, yielding a decay time between about one and six seconds, as well as some timbral changes. The bass cut control acts on the drive circuit to reduce low‑frequency rumbles, while Treble is a high‑shelf cut/boost filter designed either to add brightness or to darken the highs. Drive models the way the original amplifiers and transducers behave at high signal levels while Analog adds in a little hum and hiss for those who want to get even closer to the original sound. Crosstalk may be faded in or out to emulate the

input cross‑channel leakage of the original design. Pre‑delay adjusts the delay between the dry signal and the plate reverb, which would once have been set up using a tape machine. One reason the four plates sound different is that the sound changes depending on the spring tension holding the plate in place, on the particular piece of metal used to make the plate, and on exactly where the transducers are mounted. Plate A sounds large and expansive, while Plate B is a little brighter and tighter‑sounding, with C moving even further in that direction. It is also noticeable that at lower

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damper settings, Plate A has a rather longer reverb time than B or C, though they all have similar decay times at their maximum settings. Plate D is most similar in character to Plate A, but with some valve‑like roundness and smoothness to the tonality. Turning the Analog knob up to maximum brings in just the slightest hint of background noise, and even that differs between plates, though I doubt any would be audible in a real world‑mix. With Analog set at maximum, the hum and noise level is the same as on the original plates. In all cases the plate sound is nailed perfectly,


whether creating a short room‑like sound or a long wash. Turning up the Drive adds a bit of grit as both the electronics and the transducers are driven harder, but it is very musical if not overdone. I set up my UA Plate 140 plug‑in as a comparison and found that it sounded rather different from any of the Waves plates. By tweaking the EQ controls it is possible to get the characters of the two plug‑ins to converge to a limited extent, but I rather like the fact that the Waves and UA products each have their own characters drawn from studios on opposite sides of the Atlantic. In general I’d say the Abbey Road circuitry mods have resulted in a warmer sound than the standard EMT design. Abbey Road Reverb Plates really comes to life when you use the plug‑in in a mix. All four models work their magic on piano, drums, vocal, electric guitar — in fact just about anything you care to throw their way. There’s a real sense of weight and three‑dimensional space, despite the fact that a real plate is essentially a two‑dimensional acoustic device. You can

almost hear all the weight of that huge metal plate. And when damped right down to one second or so, plates B and C sound lovely on drums or percussion if you just want to add life without a big wash of reverb. This is one product that probably doesn’t need a bypass button, and could well end up being your go‑to reverb for just about any pop or rock production work. Now if only Waves would give their software engineers a long holiday to give us a chance to catch up with reviews of all their other new plug‑ins! Paul White $$ $99. WW

Eventide T‑Verb Formats: Mac & PC VST, RTAS & AAX; Mac AU

I recall interviewing Tony Visconti many years ago, when he was still operating Good Earth Studios in Soho, and he explained to me the setup of gates and room mics that this plug‑in emulates. We were discussing the vocal treatment he used on the David Bowie track ‘Heroes’, from the album of the same name. At July 2016 / w w w . s o u n d o n s o u n d . c o m

the time, it was a very inventive way of working around the limitation of having only one tape track left for the vocal. It was also quite brave, as the gate settings that are a key part of his original setup are crucial, and a wrong setting could have ruined an otherwise perfect take. Tony set up three microphones in the hall of Berlin’s Hansa Studios: one conventional close mic and the other two at different distances back in the hall, where they would pick up a lot of the hall’s natural reverb. Gates on the second and third microphones were set to open at progressively higher thresholds, so the quietest vocal passages would see only the close mic go to tape, while the loudest would see a blend of all three recorded. Eventide’s take on the idea requires a bit less commitment from the engineer! T‑Verb incorporates three completely independent reverbs plus compression and selectable polar patterns on the close vocal mic. The gates on microphones two and three are fully adjustable, with


meters for level and gain reduction as well as gain, threshold, attack, release and hold controls. While it is possible to emulate the original ‘Heroes’ treatment (which is included as a preset), T‑Verb would be a bit of a one‑trick pony if that was all it could do. In fact, the plug‑in goes rather further than the original: the room mics are stereo and the mic positions can actually be automated along with the other parameters, to make it sound as though they are being moved around the room.

We’re not told exactly how the reverb is generated, but the emulation of Hansa’s Studio 2, the Meistersaal concert hall, sounds extremely believable, and Eventide have managed this without imposing too much of a CPU load. The two far microphones can be placed anywhere in the hall, and the overall room reverb can be adjusted for EQ, diffusion and decay from the master Room section. The two gates come post‑reverb, with console‑style faders controlling levels and

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stereo placement. There are also solo, mute and polarity buttons to optimise or exploit the phase relationships between the three mics. Most of the settings are self‑evident, though the near mic settings only appear when you hover the cursor over the mic graphic. This gives the option of engaging separate high‑ and low‑cut filters and switching between omni, cardioid and figure‑of‑eight polar patterns. As you’d expect, a number of presets


have been included, and a useful Mix Lock feature allows the user to audition them while keeping the wet/dry mix constant. As the basic reverbs are so good, the plug‑in could simply be used to produce a hall or chamber ambience, and it will do so very well, with plenty of variation to be had just by moving the mics and tweaking the decay and density controls. However, those gates can be used in a number of ways to make the processing more interesting. As the gates have separate Hold controls, they can be used to set up very convincing gated‑reverb treatments, which sound rather more authentic than the faked ER bursts so often used in place of gated reverb. Then there are variations on Tony Visconti’s original setup to produce reverbs that come into play only when the input is loud enough. The wealth of presets, created by Tony himself and by a number of guest producer/ engineers, really shows off the scope of this plug‑in, which I found worked particularly well on drums and percussion, though it can be applied to pretty

much any instrument or voice. When I first heard about this plug‑in I wasn’t sure quite what to expect, as doing the ‘Heroes’ thing is obviously a bit specialised, but having tried a number of different instruments and settings, I’m really starting to appreciate what T‑Verb can do — and it’s a lot. You can get a free demo so give it a spin. I think you’ll like it. Paul White $$ $149. WW

PSP B‑Scanner Formats: Mac & PC VST, RTAS & AAX; Mac AU

Hammond created their so‑called ‘scanner’ chorus/ vibrato circuit using a hybrid of electronic circuitry and mechanical switching. At its heart is a nine‑stage LC phase‑shift/delay line similar to what you might find in an analogue phaser pedal; but the delay time is modulated by a rotor arm switching between the outputs from the individual stages using a capacitive non‑contact system. This has the effect of crossfading the nine delay steps rather than switching July 2016 / w w w . s o u n d o n s o u n d . c o m

abruptly from one to the other, though the effect is still slightly ‘lumpy’. By cycling upwards and then back again, 16 steps are used to create each modulation cycle. Blending in some of the dry signal makes a chorus‑like effect possible, as well as pitch vibrato. PSP’s B‑Scanner plug‑in replicates the vibrato, but offers far more scope to create effects outside the range of Hammond’s original design. Looking very similar to their L’otary rotary speaker, the B‑Scanner interface is divided into colour‑coded panels, with three main sections including separate fast and slow speed settings with ramp‑up/ down inertia. A rotary display shows the scanner activity in green and red for the two channels, with yellow showing when they are in phase. Random modulation to create variations between the channels is available to emulate worn (or broken?) hardware. A Mode Selector engages one of six emulation modes, while the speed section is somewhat like that of a rotary speaker with fast,


slow and stop options, but also with the ability to set different speeds for the left and right channels. The speed lever may be used as a continually variable control or as a switch. HF EQ is also provided along with a ‘HiRes’ option that uses a 25‑stage, 48‑step emulation rather than the nine stages in the original. And if this still isn’t smooth enough, there’s a completely smooth option that strays even further from the original. Amplitude modulation or tremolo can also be dialled in and the wet/dry mix is variable. Other niceties include a stereo Spread control, Drive to add some amp saturation, and tempo sync’ing. Sonically, the ‘original’ presets create the charming lumpiness of the original, while choosing a smoother setting creates a more polite result. It is possible to get close to a rotary‑speaker sound but without split‑band modulation. There’s definitely something endearingly retro about the scanner sound, which PSP have emulated very faithfully, and it makes a useful alternative to the traditional Leslie sound, as

the tonal changes caused by the Leslie cabinet are absent and the modulation character is sufficiently distinctive from standard chorus and vibrato. It also works well on a range of July 2016 / w w w . s o u n d o n s o u n d . c o m

instruments from organs and synths to guitars and voices. In all, a very useful modulation tool. Paul White $$ $69. WW

Sylvia Massy:

Adventure Recording

Internet connection required.


n January the SOS Team took a trip to Ashland, a small town seated at the bottom of a snow-covered mountain in Oregon. The town itself is home to producer, engineer, mixer and author, Sylvia Massy. Known for her work with acts such as Prince, Tool, Johnny Cash and System of a Down, Sylvia has carved something of a niche in the industry as a lover of adventure recording: that is, unusual techniques that go beyond the ordinary realms of music production. We get some top tips for going off-piste in the studio. July 2016 / w w w . s o u n d o n s o u n d . c o m


Elysia Karacter 500

Analogue Saturation Processor Analogue distortion devices abound, but few offer as wide a range of sounds as this one. M AT T H O U G H T O N


t seems that the cleaner we’re able to record, the more obsessive becomes our hunt for ways in which to add back in a bit of ‘analogue mojo’. To that end, there’s a small but growing market of

high-quality analogue ‘saturation’ devices, of which Elysia’s Karackter and Karacter 500 are the latest additions. It’s the latter which I’m reviewing here; the 19-inch rackmount version, announced shortly before we went to press, seems to be almost identical in terms of functionality

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and controls but it adds CV inputs, enabling you to hook up an external control source, such as an LFO, to change the effect over time.

Overview Saturation seems a bit of a vague term these days. For me it conjures up sonic memories of overloaded analogue tape and transformers, but what we have here is a solid-state analogue device designed to add a more tube-like harmonic distortion. While the effect can be subtle, it doesn’t have to be so: during the review period I created a number of audio examples and if you listen to those, as well as the ones on Elysia’s site, you’ll be able to hear that this can be suitable for anything from the very faintest enhancement of a vocal to full-on loop-mangling distortion. This device differs from so many guitar stompbox effects in two key ways. The first is the degree of control it affords the user, which I have to say is remarkable, and the other is the quality of the signal path — other than for the distortion you dial in, it sounds remarkably clean. Indeed, Elysia describe it as “mastering-grade saturation”; I suspect the point they’re making is more about the quality of this processor than its intended applications. We’re all taught not to judge a book by its cover, but first impressions count and, for a 500-series module, this device cuts a stylish figure. It sports the same slick livery as most of Elysia’s range: a matte dark blue with white legending, distinctive and tactile detented rotary controls, and buttons (with small, bright, recessed LEDs) to engage the various functions. The white backlit logo, recessed in a bevel-edged circle, adds a pleasing finishing touch; it all

Elysia Karacter 500 $975 PROS

• Great build quality and stylish appearance. • Mix control makes subtle or abusive results easy to refine. • M-S stereo mode. • FET Shred and Turbo Boost modes inject real attitude. CONS

• What cons? SUMMARY

This is a classy, intuitive and supremely controllable distortion processor, capable of sounds from the subtle to the extreme.

looks and feels very classy. But the most important thing about the front panel is not its aesthetic appeal, but the fact that everything’s laid out such that the controls are easy to access and easy to read; you’ll have absolutely no trouble getting your fingers on the controls or figuring out how to drive this thing.

On Test I chose to begin by deploying the Karacter on various parts of a multitrack acoustic drum recording and discovered, to my delight, that it’s as suited to a gentle enhancement role on the stereo drum bus as it is to more creative applications with individual kit pieces. This is largely thanks to its Mix control, as no matter how aggressive the processing, you can always turn it anti-clockwise to back off the effect. In terms of general approach, I found it best to start with the Mix control set to 100 percent wet and tweaking the Drive and

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Color controls until arriving at a desirable sound character. Then, I’d adjust the Gain knob to match the level of the processed and unprocessed signals, using the bypass button to do so. Working this way makes the Mix control far more intuitive, as it gives you a 50:50 balance at the 12 o’clock position. It’s then a simple matter of nudging that balance to taste. If you’re using the Karacter for more ambitious mangling of individual kit pieces (be they acoustic or electronic) it’s worth noting that the more aggressive distortion will inevitably weaken the bottom end of the drum sound, something that can rob both kicks and snares of their ‘oomph’. The best way to counter this, I found, was to over-egg the processing a little before blending back in more of the dry sound, which of course has its low end still intact. As long as you pay attention to such things, the results can bring a real smile to your face — they certainly did to mine when I transmogrified an acoustic kick into the sort of fat, squelchy monstrosity that you’d think could only have originated from a drum machine. It lends itself equally well to snare distortion. And as there are two independently controllable channels, of course, you could process both kit pieces differently with the same unit to radically refashion a drum part, without mashing its cymbals into oblivion. So it’s a versatile processor for drums, then, but what of other sources? On a cleanish bass guitar part it proved useful for adding a judicious helping of mid-range ‘bite’ without things becoming overblown or ill-defined, and I also found that the Karacter 500 could work very nicely as a parallel distortion device for vocals, and for male rock- and rap-style parts in particular. For electric guitar, you’ll want to try

Elysten For Yourself! I’ve prepared a few audio examples for your delectation (http//sosm. ag/jul16media), which are intended to complement the ones on Elysia’s web site ( karacter-500/introduction). They won’t reveal everything that this unit can do but between them they should give you a clear idea of the sounds that are achievable. a control I’ve not discussed yet: FET Shred, which enacts another solid-state distortion circuit. The idea, Elysia say, is to mimic the sound of a cranked tube amp. The associated Turbo Boost button shifts the threshold for this circuit a little lower, giving you a stronger effect. If you were hoping for an accurate model of a tube amp, you might be disappointed, but it is certainly reminiscent of one, and can be a very enticing sound in its own right. That Mix control again means that you have the ability to dial in as much or as little attitude as you wish. Whereas it’s always essential to tweak the Color, Drive and Gain controls, you’re not going to want to use FET Shred (with or without Turbo) on every source, but it’s by no means limited to use on guitar: on the right part, it can sound great on drums, bass, synth pads, arpeggios and more.

Stereo Sources Although the Karacter 500 is a stereo/ dual‑mono processor, I found myself using it more on individual sources than on stereo buses, and that’s probably because that gives you two ‘instances’ to use in your mix. But Elysia have certainly put a lot of thought into the stereo side of things.

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In dual-mono mode not only can you use the channels on different sources, but you can apply different treatments to the left and right channels of a single stereo source. That can be useful when the material has different things going on in the left and right, of course, but it also presents the opportunity for more creative processing. You can, for instance, add some real drama to a plain-sounding stereo synth, guitar or organ part, or add a sense of width to a mono source by treating the same part differently in each channel and oppositepanning the results. That’s something that worked well for me with a guitar part: it’s effectively re‑amping the part twice with different settings, to spread the sound across the stereo panorama. With stereo linking enabled, the left-hand set of controls governs the behaviour of both channels at once, and in the M-S mode one set of controls is used to process the Mid signal and the other the Sides. This means you’re able to do some remarkable sound-design work on electronic drum loops, synth arpeggios and the like, targeting centrally panned elements and treating them differently from those panned to one side. It also makes this a useful device for processing stereo drum samples (eg. kick, snare, hats, claps or more complex loops), for which the main energy you wish to reshape is in the middle but there’s plenty of ambience in the sides that needs room to breathe.

Konclusions Really, this is one of those devices that, ideally, you’d take for a test drive, to discover whether it offers a sound you like. For me, it proved versatile, easy to use, sonically satisfying and enormously

Alternatives So many devices can be driven into saturation that it’s impossible to list them here, but a few offer a wide range of sounds, or afford the user a particularly useful degree of control. These include the DIYRE Colour system, the Overstayer Saturator, various devices made by Looptrotter and some electronic analogue tape emulations, such as the Sound Skulptor STS, Roger Mayer 456 Stereo, and Rupert Neve Designs Portico 542. These all offer things that the Karacter doesn’t — but then none offer quite what this does either!

fun. The settings you end up with for different sources might vary quite radically, according to both the nature of the sound (how short, sustained or rhythmic it is, how much low- or high-frequency information it includes, its level, any effects tails that are printed, the amount of leakage from adjacent sources during recording, and so on), but arriving at those settings is always intuitive, and there’s bags of control over the result — more than most saturation devices are blessed with and certainly more than an overdriven preamp. I could no doubt compile a wishlist of additional features (high- and low-pass filters for the wet signal, for example), but most of those things would add to the price or complexity, or would make it fiddlier to use — and they can already be done by using the Karacter in tandem with other devices. As things stand, I really can’t fault it, and the versatility in terms of channel configurations is a particularly big plus too. I was sad when the time came to hand this unit back! $$ $975. EE WW

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WIN! Meris Mercury7 500-series reverbs Worth



he Meris Mercury7 — named for NASA’s seven Mercury mission astronauts — brings back tactile control and inspiration to a reverb world now dominated by pixel interfaces. The device is a mono hardware DSP reverb that fits into a 500-series rack. It sports two distinct proprietary algorithms, ultraPlate and Cathedra, both of which were born out of love and admiration for sci-fi soundtracks of the ‘70s and ‘80s. The modulated reverb tails of Vangelis’ Blade Runner score in particular were a sonic benchmark for the Meris engineering team throughout development. A single ‘Modulate’ control varies a complex matrix of parameters behind the scenes to accomplish some of the most ethereal modulated reverb sounds

but exemplary algorithmic mono reverb” adding that it produced “genuinely wonderful virtual spaces with a massive range of characters as well as some really quite weird and wacky special effects.” While the sonics were inspired by a vintage vision of the future, the Mercury7 hardware is the best of the best by today’s standards. Premium 24‑bit A-D/D-A conversion and 32-bit DSP ensures that the processed signal enters and exits at the highest resolution and quality. On top of that, an analogue mix bus at the output ensures that the dry signal is never converted to digital, so that its latency is zero, and that the entire signal path has an exceedingly low noise floor. For stereo or surround effects busses, parameter synchronisation

available today. High and Low damping controls vary the regeneration energy of the high- and low-frequency reverb content, providing flexible tone shaping of the wet signal. A Swell envelope switch offers a dreamy, slow‑attack ‘wash’, while the unique intra‑tank pitch regeneration takes the effect to new heights with shimmer, detuned, or octave‑shift reverberation. SOS Technical Editor Hugh Robjohns was so taken with the Mercury7 that he was reluctant to return the review units. In his June 2016 review, he called it “an unusual

is provided with simple internal linking cables. In order to be in with a chance of taking this pair of Mercury7 reverb units (with stereo linking cable) home to your 500‑series rack, simply follow the big red link to the competition page on our web site, and fill in the form before Monday 8th August, 2016.

Prizes kindly donated by Meris. TT +1 747 233 1440 WW

TO ENTER, PLEASE VISIT: July 2016 / w w w . s o u n d o n s o u n d . c o m

Emulation Impossible Software may attempt to emulate some of the character of an analog console, but nothing can capture the vast soundstage, articulation and headroom of our unrivaled analog summing circuits paired with the ultimate multi-channel D/A converters. Your Digital Workflow • Stunning Analog Sound



-Powerful yet musical D to A conversion -Precise ultra-low jitter clock

-Unsurpassed analog summing -3 custom color circuits



Hypersynth Xenophone Analogue Synthesizer In a world of reissues and recreations, the Hypersynth Xenophone is resolutely doing its own thing. PAUL NAGLE


e’ve been served so many variations on the theme of ‘subtractive analogue’ that it’s tempting to assume there’s nothing left to say. The Xenophone is a monophonic/ duophonic synthesizer from mystical Persia and it aims to challenge that assumption — and a few more besides. Armed with discrete circuitry, the Xenophone maintains an analogue signal path right up to its 24‑bit effects. It offers full MIDI control of every parameter, plus a smattering of CV too, and is powered by three intriguing DCOs

and two slightly unusual sub oscillators. Hypersynth’s tabletop box boasts a sizzling multimode filter, software envelopes and LFOs, plus an arpeggiator that can impersonate a multi‑lane step sequencer. PC users are offered a free editor that runs stand‑alone or as a VST, but if you have a Mac, you’ll need to be patient a while yet (Q3 2016 is the rough estimate). Those who prefer the more traditional hands‑on control are invited to grapple with hardware and more buttons, encoders and LEDs than many larger synths can boast. Which, as it happens, is exactly our kind of invitation!

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Arabian Lights Nobody could accuse Hypersynth of skimping or failing to fill the panel space. With the exception of MIDI reception status, every conceivable function has an LED. That’s 88 of them — all potentially dazzling! Fortunately, they are of variable intensity and ultimately less troublesome to the eye than the tiny ‘select’ buttons, whose black‑on‑black colour scheme was completely lost in my gloomy man‑cave. As well as a host of dedicated controls, you’ll find four multi‑purpose encoders (E1‑E4) positioned beneath the 2x20 blue OLED display. The encoders are utilised by the active envelope or by other functions for which no dedicated encoder or switch exists. Personally, I found the sharing of envelope controls far from ideal, probably due to my habit of toggling between filter and amp envelopes repeatedly. If you, too, are in danger of wearing out your button finger, I recommend studying the downloadable manual and the alternate way of getting around — the menu system. Fellow envelope fetishists will appreciate one shortcut in particular: the Xenophone remembers the last envelope edited. Therefore, at any time you can switch from tweaking another area and return to it with a single button press. Surely the most puzzling aspect of the interface relates to the way the E1‑E4 encoders are employed. The general rule is actually quite simple: if a dedicated encoder or button exists, you must use it. Adapting to this is seemed unnatural at first though. For example, supposing you adjust the filter cutoff frequency, the action will update the display to show the four values relevant to the filter section — ie. envelope depth,

Hypersynth Xenophone $999 PROS

• An analogue synthesizer with extra waveform choices and a unique personality. • Operates in monophonic or duophonic modes. • The filter is lovely. • Has analogue distortion and DSP effects. • Its arpeggiator can act as a step sequencer and modulation source. • Generous patch storage. • Free patch editor with DAW integration. CONS

• Not cheap. • Some design choices feel awkward. • No Mac/AU version of the editor yet. SUMMARY

A powerful high‑end monophonic synthesizer with a neat duophonic mode. With its 24‑bit effects, rich filter and flexible arpeggiator, the Xenophone is analogue but different.

cutoff, resonance and keytracking. However, of the four soft encoders, only E4 responds to your touch — because only keytracking has no panel control. Once you get over this, you very quickly learn the position of every dedicated control. The encoders are high‑resolution, multi‑turn types, but those of the review model were solid black rather than the cool, grey‑ringed examples on Hypersynth’s web site. The filter is of 12‑bit resolution (using NRPNs) and divided into a whopping 4096 steps rather than the 128 available to a regular 7‑bit CC. Little stepping is evident — other than from the bi‑polar filter

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envelope depth encoder. If you switch on the transmission of MIDI CCs & NRPNs, you can capture complete performances into your DAW or hardware sequencer. If you like a bit of wood with your analogue, the Xenophone can be supported by two types of wooden end‑cheeks. It’s up to you whether to order the stylish angled versions (that lift the synth from the desktop) or opt for the standard flat look. On the rear of the synth are the balanced stereo outputs, audio input and headphone jack, plus a single stereo jack catering (in a way) for CV/Gate In/Out. A main volume knob is tucked away back there too, joined by the USB port and MIDI In and Out sockets. Predictably, the power supply is external but I was relieved to learn the temperamental version supplied for review has now been superseded.

Hyper Active Armed with seven memory banks, the Xenophone can hold a total of 896 sounds! Unsurprisingly, most of the banks are initially empty, but a tour through the populated quarters reveals a synth with biting, floor‑wobbling oscillators, clear highs and a squelchy, mud‑through‑your‑toes kind of filter. With intelligent use made of the sequencer, analogue distortion and other effects, the Xenophone has a surprisingly ‘produced’ sound out of the box. The experience left me eager to plough ahead and make noises of my own. Starting with the two main DCOs, they’re a sophisticated pair offering familiar and unfamiliar analogue waveforms over a massive 10‑octave range. Unusually, each offers phase‑locking and full control of starting phase — desirable attributes when designing precision bass lines requiring

consistent energy. Several waveforms consist of combinations (eg. sawtooth and square); these are blended by a waveshaper circuit, resulting in extended tonality if compared to a simple mixer. Both the blend amount and the pulse width of the square wave component can be freely modulated. Already this throws up fresh tonal avenues to explore but for weirder paths still, I recommend the catchily‑named Xor1‑2. Only found in DCO1, this consists of the ring‑modulated output of square waves plucked from both main DCOs. It’s dirty, limited and reminiscent of the Korg MS20’s ring mod, but it’s a fine example of squeezing extra juice from regular analogue waveforms. DCO1 and 2 are boosted by sub oscillators that continue the theme of innovation. As expected, they deliver square waves at one or two octaves below the main but in addition, they indulge in a spot of ring modulation. Each involves the primary oscillator and a sub two or four octaves down but, fortunately, you needn’t know what’s involved to appreciate the hard‑edged and buzzy waveforms. While not instantly life‑changing, I found them valuable additions as I got to grips with the intricacies of Xenophone programming. There’s a third oscillator (of lesser stature) too. DCO3 produces either a basic square wave or noise. Although simple, it offers a full 10 octaves of shift — and noise connoisseurs will enjoy the white, pink and roaring red output before flipping over the crunchy, pitch‑trackable ‘C=64’ noise. (It’s inspired by the SID chip from the Commodore 64.) Navigating to the Voice menu, you’ll see an option rare in DCO synths: FM. The implementation is slightly restricted in that

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you can only modulate DCO1 with a triangle wave extracted from DCO2. The manual advises that for ‘best results’ DCO1 should be a square wave, and it’s right because the results are strangely subdued with other waves. In its favour, high levels of FM will trash DCO1’s sub oscillator in freaky and unexpected ways. The Voice menu holds more goodies: Detune and Analog Drift. The former skews the pitch of the two main oscillators in opposite directions leaving the third unshifted. Drift nudges all oscillators away from the strict tuning imposed by the Xenophone’s microprocessor. At their most thoroughly nudged, they become

very un‑DCO‑like indeed. Portamento is available in fixed mode or is scaled according to the distance between played notes. The Xenophone is therefore equally applicable to prog‑rock wankery or slidey 303 bass lines. Last but not least, DCO1 can be hard‑synchronised to DCO2. Sync is a source of spiky new tonalities but, in keeping with the Xenophone’s unusual nature, the slaved oscillator all but disappears when the two oscillators are the same pitch. Interestingly, the sync operation works very smoothly on a DCO slave, but produces audible steps on the sub‑oscillator component.


The oscillators aren’t the full extent of the available audio palette; joining them in the mixer is another ring modulator (a regular one that isn’t fixated on square waves) and an external feedback loop. If you insert a lead into the audio input — eg. to process another synth or drum machine — the feedback loop is interrupted. Otherwise, feedback level becomes a significant contributor to the Xenophone’s audio shaping and is closely related to the current filter mode. With so many possible sources, the mixer is easily overdriven even before you boost the feedback or resonance. I was seriously impressed by the scope for howling,

saturated tones by the simple act of varying the input levels.

The Filter After breezing through the demo patches, I was already seriously intrigued by the multimode filter. Hypersynth’s acidic design is notable for maintaining its power even when you wind up the resonance. It includes the familiar low‑pass, high‑pass, band‑pass and notch modes, and there are three slopes available (12, 18 and 24 dB) to the low‑pass and high‑pass. While all were brimming with analogue vitality, I found the band‑pass mode the least appealing, but even that was saved by


ramping up the feedback and resonance, exposing its harder edge. The notch filter is a fine example of the breed and with high resonance and a touch of feedback it rips like a B‑movie masked maniac. Rounding off the available modes are a couple of low‑pass filters placed before a 6dB high‑pass — they cut through like a rusty scalpel. At high resonance, most filter modes are capable of self‑oscillation, but in these serial modes, the resulting waveforms are a triangle and clipped triangle wave

respectively, rather than a regular sine. Now seemed the opportune moment to prod the filter FM button, enabling DCO1 as a modulation source. The button removes the oscillator from the audio path but its presence is still keenly felt, especially on a self‑oscillating filter. My afternoon faded into evening in a blaze of FM‑based percussion, with killer results obtained from even basic connections such as velocity controlling pitch or the modulation envelope driving feedback.


PC users are offered XEditor, a free stand‑alone editor and VST plug‑in to reveal every corner of this Aladdin’s cave. I’d love to offer my thoughts on it but unfortunately there’s no Mac version currently available. The Xenophone is not a class‑compliant

device either, so you’ll need to digest the instructions on the Hypersynth web site before using the USB MIDI connectivity and performing firmware updates. Fortunately, although the latter process is more complex than usual, it’s well‑documented.

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In less than half a day, I created almost 50 patches of the aggressive, slicing, tearing persuasion and felt I was homing in on the Xenophone’s sweet spot. For me it excels at dark, acidic bass, screaming solos and every strain of wonky, glitchy cross mod and ring mod. It’s also very capable of generating classic mellow, brassy leads if you’re that way inclined. Incidentally, here’s a tip I really took to heart after overwriting several patches by accident. The default ‘save location’ is the last one specified. However, if you turn the E1 encoder prior to saving, the destination becomes the current source instead.

Pushing The Envelope Ordinarily, you might take modulation for granted. With a trio of envelopes and LFOs, plus a modulation matrix to hook everything together, you might feel there’s nothing to discover, but that would be to miss a treat. Several, actually. The DAHDSR envelopes deserve attention, and not just because of those extra hold

stages. Other options are but a menu page away, and they’re options rarely seen and even more rarely appreciated. For a start there are four styles to choose from, essentially choices of envelope shape. These are: linear, exponential and a pair of ‘RC’ shapes. The last two are aimed at replicating the distinctive curve of capacitor charge and discharge found in vintage synths — great stuff! Better still, you can independently select the way each envelope is triggered. The options are to reset the envelope level on every key‑press or to maintain the level only when you play legato, but it’s the final two modes that hit pay dirt. In either of these ‘analogue’ modes, the envelope acts like a classic single or multi‑triggering synth. The crucial difference between this and other behaviours is that the envelope level is not constantly reset to zero. Not only is this essential to the enjoyment of long attack and release values, it neatly avoids the clicks that invariably accompany the ‘zero‑ing’ type of envelope. (Owners of Korg’s

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Minilogue will know what I’m talking about.) With envelope times ranging between 1ms and almost 30s and built‑in velocity control for the amp envelope, there’s much to applaud, but there’s more. Each envelope can be restarted by LFO2 (with the weird condition of sample & hold being selected) or by MIDI CC64. Of more practical value, the envelopes’ attack and decay stages are loopable, and with such fast envelopes, a looping envelope becomes another audio‑level modulation source. Finally, if you take release to its maximum, the envelope enters a special hold mode. This is ideal for processing external signals by the filters and effects as it bypasses the requirement to keep hitting a key.

More Modulation With three LFOs to play with, it’s no great hardship that LFO 1 and 2 share controls and targets, but some other limitations are less palatable. For example, only LFO1 and LFO3 have built‑in (although menu‑bound) depth controls. Encoders and buttons are allocated to speed, waveform and target selection, but if you want any modulation it involves entering a menu. All the usual waveforms are offered and each LFO can run freely, sync to the current tempo or be restarted by playing a note. Their range is from stopped to just below 100Hz. In order to make use of LFO2 or the third envelope, you’ll need one of the eight

precious modulation slots. A slot consists of a single connection between any of the 40 sources and an impressive number of targets. In practice, it doesn’t take long to use them all up — the LEDs are a helpful reminder of which are taken. Amongst the potential sources are several useful combinations, such as LFOs that are multiplied by the value of the mod wheel, by each other or by the mod envelope. For further versatility, modulation targets include the envelope stages, the depths of the first four matrix slots and even some effect parameters. The modulation matrix is also where you go to assign incoming or outgoing CV. Examples of this are to direct one of the Xenophone’s internal modulation sources to another synth, or to draw modulation in from an external source. Although it is possible to use the Xenophone as a MIDI to CV interface (I successfully drove my faithful Roland SH101 from it), you can’t play it — in tune anyway — from an external CV/Gate synth or sequencer. I’m informed this is due to a limitation of the DCO design.

Arpeggiator/Sequencer The arpeggiator has no immediately obvious panel control to switch it on or off. Instead you press the relevant Select button until you reach the menu page with ‘Span’ on it. Span’s options are: Off, Up, Down, Up/Down, Step and Ordered,

MIDI Every aspect of the Xenophone may be harnessed by MIDI. This includes the four sequencer rows, effect parameters and even the modulation matrix entries. Some of the MIDI implementation is

a little quirky, eg. CC10 (which you’d expect to control pan) is assigned to DCO1’s phase but CC8 (balance) is employed to move the output — effects and all — around the stereo spectrum.

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and when you pick anything but ‘Off’, the arpeggiator kicks into life. There’s no ‘random’ direction but when you pick ‘Step’, the arpeggiator becomes a step sequencer, it patterns up to 16 steps. When in sequencer mode, the pattern plays as soon as you play a note or, providing Latch isn’t enabled, you can stop, preview and pause using the mini transport keys. Each sequence step contains a note and velocity — or simply a rest — plus two ‘auxiliary’ values. The auxiliaries are unassigned at first but, since all four rows are available sources in the modulation matrix, they are powerful tools for introducing movement and modulation. To exploit this idea further, the sequencer can be flipped to ‘Step LFO’ mode, in which pitch and velocity control is disabled, leaving all four rows for any purpose you like. I said those eight matrix entries were quickly used up! Hold can be added to any step, and slew introduced to smooth the otherwise abrupt transitions from step to step. Sequences can be transposed by keyboard input and notes

entered from a keyboard too, unless you prefer step‑by‑step entry from a menu.

Effects Not many analogue monophonic synths have built‑in effects and typically, when they do, it’s a basic delay of some kind. Hypersynth’s effects consist of analogue distortion plus a 24‑bit digital reverb and delay. There’s no balance of distortion to signal, just an on/off button, so it’s a challenge to introduce subtly. Distortion has a crude transistor feel and goes from ‘light’ to ‘massive’ in a mere four button presses, each adding volume, body and a progressively harder‑edged distortion. The effects include five reverbs (room, hall, cathedral, gated and plate), plus two delays and a combined delay and reverb. Some algorithms are more tweakable than others but who’d deny the sheer usefulness of built‑in effects, especially for live performance? The reverbs are particularly good, especially the vast and magnificent cathedral. The parameters (pre‑delay, time and damping) are assigned arbitrary values


in the range 0‑200 but they aren’t hard to master. With a maximum time of 1 second and a pleasant ping‑pong mode, the delays are somehow less impressive but still handy, especially in tempo‑synced mode.

Duophony A software update arrived as I was almost wrapping up. It contained several important bug fixes (shortening the review somewhat) and a sweetener in the form of a new duophonic mode (lengthening it right here). Naturally this has a twist; when ‘Duo’ is enabled, ordinary monophonic playing acts on DCO1 alone, but when you play a second note this brings in DCO2 and 3 together. The approach feels more natural than duophonic implementations in which all oscillators play until you hit that second note, when they are divided. However, perhaps the most important characteristic of the implementation is the drafting in of two separate amp envelopes and VCAs to shape the voices before they hit the filter. Each is individually velocity responsive. This presents the possibility to play and hold a note that will decay naturally while playing a solo over the top. Of course, you’re still limited to a single filter, but as each oscillator may have different waveforms and be modulated separately, drone and solo combinations work better than on many duophonic synths.

Conclusion Digging into the Xenophone’s Greek roots, they translate to either ‘different’ or ‘foreign’ plus ‘voice’ or ‘sound’. By my reckoning this nails it. The Xenophone is indeed different enough to offset potential analogue fatigue. It’s a refreshing break from Moog‑like traditions and thanks to those extra waveform choices, FM of filter and oscillator

Alternatives The less expensive but more menu‑bound Waldorf Pulse 2 springs to mind as competition. Despite its minimal interface, the Pulse 2’s triple DCO design, paraphony and extensive modulation matrix offer considerable flexibility, but it lacks the effects, step sequencer and sweet filter modes of the Xenophone. Other contenders include the Studio Electronics Boomstar range: they’re based on the oscillators and filters of classic synths of old but are comparably priced and can be driven by CV/Gate. The Boomstar’s direct, knobby interface is in stark contrast to the encoders, MIDI control and patch memories of the Xenophone.

and the chewy multi‑mode filter, it deserves to win many friends. Although DCO synths are sometimes viewed as less ‘alive’ than VCOs, that opinion seems increasingly stale in the light of synths like this which can deliver fat, trans‑Atlantic basses and swooshy leads galore, optionally bathed in lush reverb and sequenced if necessary. I’m sorry I couldn’t try the XEditor plug‑in but, despite its menus, the Xenophone is accessible enough anyway. However, it is not a budget synth nor is it necessarily the easiest to warm to. As my personal preference tends to be knobs over encoders, I admit my first days were marked with frustration over aspects of the encoder behaviour. Fortunately, this feeling soon passed, but Hypersynth say they are considering opening up all the soft encoders in a future update. And speaking of updates, duophony was a late surprise, but a very welcome one! If you like a little spice and mystery with your analogue, the Xenophone could prove irresistibly moreish. $$ $999 WW

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Acustica Audio White

Equaliser Plug-in Acustica Audio’s new Acqua series plug-ins are painstakingly ‘sampled’ from high-end hardware. Does the end justify the means? SAM INGLIS


hen software developers aim to capture the sound of studio hardware in plug-in format, they’ll usually employ some sort of algorithmic modelling technique to do so. In theory, this has quite a few advantages. Modelling plug-ins typically require relatively little memory or drive space; it’s often possible to optimise the code so that they run very efficiently; and it’s possible to add additional features that aren’t available on

the original. But do they fully capture what’s special about the sound of the hardware? Quite a few engineers still feel that the answer to this question is ‘No’, which is why Acustica Audio’s Nebula has such a devoted following. Nebula uses an alternative technique called dynamic convolution. Basic convolution, as used in most convolution reverbs, is carried out by sending a single impulse response through a piece of equipment or an acoustic space and capturing the results. As such, it assumes the same behaviour at all signal levels, so it’s not

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capable of capturing non-linear effects such as saturation. Dynamic convolution overcomes this limitation by making lots of measurements at different signal levels, and interpolating between them. You could draw an analogy with sampling: standard convolution is like taking a single sample of an instrument, while dynamic convolution is akin to multisampling at different velocity levels. Dynamic convolution conspicuously lacks the above-mentioned advantages of modelling. It places a heavy burden on the host computer’s CPU and memory resources, and as you multiply the number of different settings that need to be ‘sampled’ — for instance on a complex piece of equipment — the necessary library of impulse responses can grow to unwieldy proportions. But if dynamic convolution has remained the province of obsessive tweakheads rather than working audio engineers, that’s also partly down to the design of Nebula. No-one denies the quality of the results that can be obtained using Acustica’s flagship program, but for many, the process of obtaining those results has just been too painful.

Crystal Waters To Acustica’s credit, they’ve recognised that if dynamic convolution is to enter the mainstream, it needs to be simpler to use. Recent versions of Nebula have improved considerably in this respect, but of course the reality is that most of us will never actually get around to making our own IR libraries, or even editing other people’s. What most people need is not a single, all-powerful convolver that takes years to master, but individual tools that sound great, are easy to use and don’t make your computer fall over just by existing. And that’s

exactly what Acustica are promising with their new Acqua series of plug-ins. In essence, what Acustica have done with each of these plug-ins is to laboriously ‘sample’ a single audio device and package the results within a custom plug-in interface. The Acqua plug-ins use Acustica’s latest Core7 engine, which supports multithreading and offers a number of advanced features ‘under the bonnet’, so although they can still impose a hefty load on your computer, they’re slicker than earlier implementations of dynamic convolution. Most importantly, from the user’s point of view, you can treat them just as you would any other plug-in: load one into a DAW insert slot, open up the graphical user interface, and move the controls around until it sounds how you want it.

The White Stuff At the time of writing, there are already no fewer than 19 plug-ins in the Acqua range, but this review will focus on the White equaliser, which is unique in being an officially licensed recreation of a hardware original — the original being, in this case, a high-end mastering equaliser designed by WSM Labs and based loosely on the classic Pultec EQP1A. The installer actually adds two separate plug-ins to your system: the standard White and a White ZL version, which operates at zero latency and is thus suitable for use when tracking. Unfortunately, there is one hurdle that has to be overcome before you can dive in. Old habits die hard, and Acustica’s new-found user-friendliness is undermined by a slightly baroque authorisation process. This involves sending a product authorisation request to Acustica’s server, waiting for their ‘global key generator’ to churn out a licence,

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and manually dropping this within all the Acustica plug-in folders on your system. It’s mildly annoying for one Acqua plug-in and would be a complete pain in the neck if you were installing many of them at once. As the screen capture that heads this review shows, the White interface definitely looks the part, with its large and beautifully rendered photorealistic controls. Looks aren’t everything, though, and I have to say that once you start engaging with the White interface, it’s rather less satisfying. On my computer, there was an irritating lag of half a second or so in response to any user input, and the impressive look of the rotary switches is undermined by the rather hesitant and vague way they move when adjusted. It’s a bit like reaching for a gorgeous Bakelite dial only to find that it’s been attached to a cheap Maplin pot!

Acustica Audio White €149 PROS

• Has that elusive quality of making everything sound better, at a wide range of settings. • Packages dynamic convolution in a conventional plug-in interface that’s easy to use. CONS

• Despite the optimisations Acustica have made, dynamic convolution still places a heavy load on your computer. • Annoying authorisation process. • User interface feels sluggish and unresponsive. SUMMARY

Acustica Audio’s Acqua range deserves to bring dynamic convolution to a much wider user base. If your computer can handle White, your ears will love it!

White Noise Acustica seem to have created a real buzz with their Acqua plug-ins and with White in particular, which has earned rave recommendations from some very well-respected engineers. Feature-wise, it follows the Pultec EQP1A in offering low- and high-shelving boost and cut (with continuous rather than stepped gain), but expands upon it in a couple of ways. Both the high and low bands have more switched frequency options than are available on a Pultec, and most significantly, it’s possible to select different frequencies for the boost and cut. White defaults to dual-mono mode, but can be stereo-linked or switched to M-S operation, where the left channel’s controls adjust the Mid frequncies and the right channel the Sides. There’s also a valve gain make-up section, which can be introduced into the signal chain even if you’re not using the EQ at all. In the past, I’ve struggled to fit Pultec-style EQs into my way of working. Master bus EQ is usually something I add towards the end of the mixing process; having got the overall frequency balance as close as possible using channel-based processing, global EQ is about providing the final touches, and a Pultec always seems too broad a brush. You can’t use an EQP1A to dip out that hint of muddiness at 180Hz, or add a tightly focused boost in the upper mid-range, or tame the boxiness that can accumulate in the 400Hz region. So, having installed White and jumped through the authorisation hoops, I found myself scratching my head as to what to do with it. To answer this question, I talked to a couple of the engineers who recommended White to me, including SOS’s own Jack Ruston. Their

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advice was to use White, or indeed any similar mastering equaliser, in a more ‘top down’ context: beginning with a rough faders-up balance, insert White over the master bus and adjust until the overall spectral balance of the mix is where you want it. You can then turn your attention to the individual channels and, all being well, you should find that the presence of the master EQ means less work is required in this department. Result: a faster and less frustrating approach to mixing, at least in theory. When you work in this way, the quality of your master bus EQ is paramount, because it’s playing a much more fundamental role in shaping the sound of your mix. And those who swear by Acustica’s White in this role definitely have a point, because it does sound very different from the other EQs I tried. More to the point, it sounds better. In particular, the effect of a high shelving boost in White is euphonic in a way that’s a little hard to describe. It’s almost as though there is some sort of compression at work in the way it seems to thicken up or fill out the frequency region that’s being boosted, without exaggerating peaks within that region. For instance, you can apply a hefty treble lift stretching all the way down to 1.5 or 2 kHz without things starting to sound thin or tinny, and without things like vocal sibilants or hi-hats poking out of the mix. If your experience of plug-in EQ on the master bus is that it has quite a narrow ‘sweet spot’, it’s refreshing to be able to sweep the White controls around all over the place and hear a variety of different good sounds, rather than one good sound and a lot of obviously wrong ones! And, of course, you don’t have to use White on the master bus. It’s equally effective at altering the sonics of individual sources

and buses, and again, the range of settings that sounds good just seems broader than is the case with most plug-in EQs. If the mood takes you, you can also employ the classic Pultec trick of cutting and boosting at the same frequency, and the provision of separate frequency settings for cutting and boosting opens up many new possibilities in this department. However, you’d need an extremely powerful computer to use White as your default channel EQ on a busy mix. I tested it on an ‘early 2014’ MacBook Air with a 1.4GHz Intel i5 CPU and 8GB RAM, and if the System Usage window in Pro Tools is to be trusted, each instance of White used some 6 percent of this memory, and between 10 and 20 percent of a single core’s CPU load, depending on how many bands were in use. You can almost feel the burden on the machine — even simple actions like dragging White to a different plug-in slot or opening the GUI are much slower than with any other plug-in I have installed. At the end of the day, though, the old adage that you get what you pay for applies just as much to system resources as it does to money. The word on the street is that here, finally, is a plug-in EQ that might just satisfy those who feel that previous efforts have never matched up to high-quality analogue units; and for many people, I think the price of a heavy CPU load and clunky interface will be one well worth paying for the way this EQ sounds. Very occasionally, when you try out a piece of equipment, your instant reaction is ‘Wow, that just sounds like a record!’ It happened to me the first time I used a Neumann U87 — and it happened to me with Acustica’s White. $$ €149 (approximately $168). WW

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Get the principal players of Berlin Strings and enter the world of a fi rst rate string quartet for fi nest, most intimate compositions. Recorded in their orchestral seating positions at the Teldex Scoring Stage, the First Chairs are an ensemble in their own right but will also add life and defi nition when layering them on top of large string arrangements. Based on CAPSULE, the most powerful articulation management system, the First Chairs Collection forms an integrated whole when using it with the Berlin Strings Main Collection. Instruments (Solo): 1st Violin, 2nd Violin, Viola, Cello. AVAILABLE AT ORCHESTRALTOOLS.COM


APS Klasik Active Nearfield Monitors

As their name suggests, these unassuming twoway loudspeakers employ a traditional, tried-and-tested design — and they pull it off surprisingly well! PAUL WHITE


or those unfamiliar with Polish company APS (standing for Audio Pro Solutions), they have been trading for 10 years and have a range of active monitors and subwoofers in their catalogue. These days there’s very little you can tell about a loudspeaker just by looking at it: outwardly the APS Klasik is just another black-box, two-way, rear-ported active monitor with few external distinguishing features. And, like many other monitors aimed at the smaller studio, the Klasik uses a seven-inch woofer for the mids and lows, with a lightweight 0.75-inch tweeter taking care of the highs. However, its claimed free-field frequency response of 35Hz to 25kHz (±2dB) is impressive, as is its 103dB SPL rating (111dB peak, per speaker at one metre). It would be difficult to find much simpler cabinets: these are square-edged, plain boxes, other than a slight rounding on the baffle edges, and the covering is a simple black plastic foil. Both drivers are recessed to sit flush on the baffle, and a discreet

green LED shows when the speakers are on. The Klasiks, which measure just 32 x 21 x 28cm and weigh 8.5kg each, are calibrated to work as a matched pair. Both power amplifiers are analogue Class-A/B designs rated at 75 Watts each, and the crossover is set at 3.3kHz. The woofer has a cellulose cone set in a rubber roll surround; the tweeter has an aluminium diaphragm and is powered by a dual-magnet system. We’re told the woofer employs a sophisticated magnetic system to promote a ‘fast’ response, though there are no technical details as to how this might differ from the norm. According to APS, the Klasik is designed to have a very smooth response both

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in amplitude and phase, as well as a highly controlled directional response, courtesy of the tweeter choice (there’s no obvious acoustical means of controlling the directivity, as the tweeter is set into a shallow plate that doesn’t really qualify as a waveguide).

Around The Back Turning to the rear panel, which has no protruding heatsink fins, the circular port is at the top centre of the cabinet and below that is a switched input sensitivity control. This is a practical option as it ensures accurate level matching between the two speakers — variable controls can be a bit hit and miss. The input may either be on balanced XLR or unbalanced phono, but there’s no balanced quarter-inch jack option. Power comes in on an IEC connector, and there are switches for power and ground lift. That leaves just the EQ controls: a tweeter level slide switch offers -1.5, 0 or +1.5 dB optoins, while the bass control has a roll-off

APS Klasik $1300 PROS

• Clean and detailed sound with excellent stereo imaging. • Good bass extension. CONS

• Their bland looks might cause you to overlook them. SUMMARY

A good-performing, mid-priced monitor that has the capacity to surprise with its revealing, three-dimensional sound.

position, a centre ‘Passive’ position, which seems to mean equivalent to what you might expect from a passive version, then an Extended position, which appears to be the default. The rolled-off version is useful for checking how mixes might sound on a small playback system. However, there’s no dedicated bass switching to compensate for the speaker position relative to room boundaries.

Listening In Testing with a range of material using the default settings reveals that these speakers are capable of presenting a very three-dimensional picture of a mix, with great clarity combined with a bass end that is solid but definitely not over-hyped to the point that it becomes boomy. As with any rear-ported speaker, you need to leave a reasonable space between the speaker and the wall behind. You can learn a lot about a ported speaker by gently tapping the speaker cone, and in this case, the sound is fairly well damped. I did, however, reset the tweeter switches to their -1.5dB position as I felt that the default July 2016 / w w w . s o u n d o n s o u n d . c o m


Alternatives The more obvious alternatives are the Adams, Eves and Focals of this world, but in reality there are many active two-way speakers that follow this general format across a range of price points.

setting was rather forward, adding a slightly artificial crispness to the sound. Naturally the ideal setting will vary depending on the room you use, but with the revised setting the speakers sounded much more neutral to my ears. Though not priced in the esoteric bracket, these are not budget speakers either. However, they hold their own within their price sector, and their vanilla looks belie

their strong and detailed performance. Their stereo imaging and sense of depth is nothing short of excellent, as is their ability to reveal detail. Importantly, if a mix has shortcomings, they won’t be backwards in coming forwards to let you know! If you’re on the lookout for a sub-£1000 pair of monitors suitable for a typical project studio room but want something that’s a step up from the usual budget suspects, these are well worth trying. $$ $1300 per pair. EE WW

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A P P W O R K S Making Music On The Move

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audio from other iOS apps so that you can use it as a signal processor, and to send audio to other apps, allowing them to process or record the output from the synth. Alongside this, there’s a MIDI Bridge that provides six 14‑bit MIDI inputs and outputs, and the two modules together provide a wealth of connectivity. The extension cabinet also includes an audio recorder. It’s not a multitrack, but you can overdub using different sounds for each layer which, if your playing and timing is good enough, allows you to build quite sophisticated pieces that you can then share as WAV files. In fact, you have to share them or lose them — the audio is stored in a buffer that’s cleared when you close the app. By the time that Model 15 was announced, I had already programmed many patches for it on my iPad Air and, although it was still being refined while I was working with it, I found the GUI to be clear and well thought out. Details such as how many fingers were needed to carry out certain operations were updated, tried, and updated again until Moog’s engineers were happy that it was as pleasant and efficient to use as possible. My only remaining gripe is that many of the modulation controls and amplifiers are scaled such that the most useful range lies between zero and just a tad more than zero, which means that you have to have very steady fingers, even when using the fine movement shortcut. Indeed, I often found myself using two attenuators to force a modulation signal down to the amplitude I wanted and, without the ones in the second cabinet, many of the sounds I was developing wouldn’t have been

possible. What’s more, I sometimes found that Model 15 lacks sufficient mixers, as do almost all pre‑populated modular synths. I know that, until people start using these synths in anger, they prefer to see more oscillators, filters, and other sexy modules installed within them, but they soon find that a lack of mixers can be rather limiting. I wonder whether we could persuade Moog to add another row

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of modules containing nothing but mixers and VCAs? In addition to the two cabinets, Model 15 offers four additional pages accessed by tabs found to the right of the console. These include a polyphonic keyboard that emulates velocity and pressure sensitivity using front‑to‑back position and by sliding your finger on the key (respectively), plus an Animoog keyboard that adds side‑to‑side expression to this. If you don’t have access to a well‑equipped master keyboard, you can use these to surprisingly good effect. There’s also a highly specified arpeggiator that, in addition to the usual octave range and direction controls, allows you to program things such as pitch offsets, velocity offsets, gate length offsets, and more. Consequently, if you just arpeggiate a single note, you can use this as an eight‑step sequencer. Finally, there’s an emulation of the Moog 1150 ribbon controller that generates two CVs with Gates, all of which you can patch into your sounds via the CP‑15B. So how does it sound? The answer, as you would hope, is that it sounds damn good. I’m not sure that it has the rawness of a genuine Model 15 (in fact, I’m fairly confident that it doesn’t), but it has a strong character that it imposes upon almost everything that it does. This may or may not be to your advantage. If you allow the synth to channel your sound design, it will guide you down the same paths as a genuine Moog modular: muscular lead synth and bass sounds, powerful effects, and fascinating arpeggiated patches. If you ask it to step beyond those boundaries, life becomes harder but it’s worth it. I spent a lot of

time persuading it to imitate orchestral instruments, polyphonic keyboards and percussion, and the results were sometimes more than satisfying. Model 15 is an instrument that will reward the time and effort you spend on it, although you may (like me) become greedy and wish that its polyphony were greater.

“Aside from the matter of only having 2GB memory, this new iPad Pro is the best iPad Apple have ever made.” Nobody can accuse Moog Music of ignoring any sector of the synthesizer market. From the most expensive synths currently on offer to some of the most affordable apps on the market, the company cover all the bases. Costing just £23 $30 (which is 0.3 percent the cost of the $10,000 Model 15 hardware, which lacks all of the extra modules and controllers), the Model 15 app is eminently affordable. For the price of a large pizza and a bottle of something fizzy, it’s far too much fun not to try. Gordon Reid $$ $29.99 WW

Apple 9.7‑Inch iPad Pro Tablet Computer


early a year and a half after the release of the iPad Air 2, Apple recently unveiled a new 9.7‑inch iPad. However, this was not the iPad Air 3, but rather a new member of the iPad Pro family.

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Physically, the 9.7‑inch iPad Pro is almost identical to the iPad Air 2, with both devices sharing the same dimensions and weight. The only subtle difference is a less‑pronounced chamfer around the bezel, presumably to accommodate the addition of the Smart Connector on the left‑hand edge. As with the larger iPad Pro, this is where you can connect a Smart Keyboard accessory designed specifically for the 9.7‑inch model, which is available separately for $149. And because this is an iPad Pro, it’s compatible with the Apple Pencil, which seemed to work just as fluidly as it did on the 12.9‑inch monster, and has a four‑speaker sound system that, while you wouldn’t use to produce music, is significantly better than before, both in terms of volume and quality. The 9.7‑inch iPad Pro’s display offers the same 2048 x 1536 resolution as the iPad Air 2, but is apparently 25 percent brighter, 40 percent less reflective, and implements the DCI‑P3 colour space (which is used for digital movie projection in the US film industry) to provide 25 percent greater colour saturation.

Apple are also debuting an impressive new display technology called True Tone. Thanks to two new ambient light sensors that measure both colour and

The Adaptor You’ve Been Waiting For Alongside the unveiling of the 9.7‑inch iPad Pro at Apple’s March Event, the company also released — some might say finally — a Lightning to USB 3 Camera Adaptor that features an extra Lightning connector to provide power to your iPad. This means that when you’re using the iPad with accessories that aren’t able to deliver power, such as certain MIDI and audio interfaces (like RME’s Babyface Pro that I reviewed in the May issue), you’ll now be able to use that accessory for

more than the length of a single charge. It might not be elegant, but it’s certainly better than the alternative! The new Lightning to USB 3 Camera Adaptor is available for $39 and is compatible with all Lightning‑based iPads. If you’re using the adaptor with the 12.9‑inch iPad Pro, you’ll get USB 3 transfer speeds; however, with the 9.7‑inch iPad Pro (and presumably all other Lightning-era iPads) only USB 2 is supported.

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brightness, True Tone matches the colour temperature of the iPad’s display to suit your surroundings, making the screen behave more like a piece of paper. True Tone is enabled by default, and, frankly, the display looked normal to me when I first started using the device. However, if you disable it (or compare against an iPad Air 2), you’ll notice the display becomes colder with an almost‑blue tint — the difference is quite remarkable. The True Tone display will be especially interesting for those running notation‑based apps, whether for editing or simply viewing, since such apps normally have a white (or near‑white) background on which the notation is displayed. With

True Tone enabled, this can make the page look warmer, as you can see from the photo I took with my iPhone 6S. In this particular instance and environment (my office), the True Tone display is noticeably beige, and it will come down to personal preference whether you leave this feature enabled or not. The 9.7‑inch iPad Pro features the same dual‑core, 64‑bit A9X system‑on‑chip as its big brother, although clocked slightly slower: 2.16 vs 2.24 GHz. This is reflected in the Geekbench 3 score, where the 9.7‑inch iPad Pro scores 5245, slightly lower than the 12.9‑inch’s 5459 score — the iPad Air 2 scored a still‑respectable 4418.

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The biggest disappointment with this new iPad Pro’s internals, however, is that it features the same 2GB of memory as the iPad Air 2, instead of the 4GB offered by the larger model. I don’t know whether this decision was made because of technical or pricing limitations (maybe Apple assume users will multitask less because of the smaller display?), but it makes this new smaller iPad Pro slightly less ‘pro’, which is a shame, particularly if you want to run a large number of instrument and effects apps simultaneously. The storage tiers for the 9.7‑inch iPad Pro are the same as for the 12.9‑inch model, meaning that the cheapest 9.7‑inch iPad Pro comes with a meagre 32GB for £499 $599. 128

and 256 GB models are available for £619 $749 and £739 $899 respectively, with WiFi+Cellular models costing an extra £100 $130 over their WiFi‑only counterparts. If your budget can stretch to it, I’d recommend the 128GB model, especially if you’re going to be recording audio or installing large, sample‑based instrument apps like Korg’s Module. One good thing about the pricing of the new iPad Pro is that it makes the iPad Air 2, which is still a very capable device, £50 $100 cheaper, although Apple have sneakily removed the 128GB model from the line‑up. If you already have an iPad Air 2, the new iPad Pro might not be a significant enough step up to warrant an upgrade. However, if you want the latest

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iPad technology and felt the 12.9‑inch beast was simply too big, aside from the matter of only having 2GB memory, this new iPad Pro is the best iPad Apple have ever made. Mark Wherry $$ From $599 WW

IK Multimedia iRig Pro Duo Audio Interface


K Multimedia’s new iRig Pro Duo sits at the very top end of the manufacturer’s extensive hardware range. It promises to be a versatile and

affordable all‑in‑one studio for iOS, Android and desktop computers with features comprising two combination preamps with phantom power, balanced line outputs (in addition to the headphone outs), direct monitoring and MIDI in/out. All the important interconnection cables are supplied in the box, including a Lightning cable for iOS, Micro USB for Android, standard USB and MIDI connection cables. The interface runs on either AA batteries or an optional DC power supply for mobile use, and draws power from a USB port on a Mac or PC. Despite being illustrated with a mic‑stand bracket in press photos, there isn’t one included. Again, IK offer an optional accessory that would be suitable for this job. Getting started with a mobile device or Mac OS is incredibly simple, since no software or drivers are required. Windows drivers are simply downloaded from the IK web site following registration. A range of plug‑in and app software is bundled with the Duo, although functionality of these offerings varies quite dramatically. By way of example, the entry‑level version of the AmpliTube amp simulator software is enough to get you going, but when switching through the presets you

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frequently run into functions you cannot access without paying for an upgrade. To compound this issue, IK’s installation bundle installs files for absolutely all of their plug‑ins on your computer — whether you own them or not — in the full range of proprietary formats. This takes up significant space, and I found myself having to go through the relevant folders and manually remove the plug‑ins I didn’t need. Such a ‘freemium’ model is quite annoying and confusing. At the other end of the spectrum however is T‑RackS Classic, a handy mastering system with RMS/average level meters, a real‑time analyser and handy multi‑band limiter and master equaliser. These plug‑ins are really fit for purpose, adding tangible value to the overall package. In session, preamp and playback quality is perfectly respectable, and the interface is simple to operate. Fit and finish is largely solid too, with smooth gain controls that aren’t easily knocked out of position once set, handy LED activity/ clip metering, and sufficient weight to the case to avoid it moving around on the desktop once everything is plugged in. Latency levels are pretty usable too, yielding good results when feeding a guitar input through an amp simulator in GarageBand on iOS, or whilst triggering a virtual instrument via the MIDI input. When recording audio I found the direct monitoring to be incredibly useful in avoiding latency altogether, but the slider switch lacked the reassuring ‘click’ of similar controls on the case. Also, in some circumstances, the audio stream goes to sleep within 20 seconds or so of playback ending — presumably designed to save power when using batteries.

This happens when playing back system audio on a mobile device and, more annoyingly, on a desktop computer whilst power is being supplied by the USB port. Since waking the audio stream can take two to three seconds, clicks and buzzes occur during this time, forcing you to return to the start of tracks playing back to monitor them without the nasty noises. This issue falls short of IK’s ‘Pro’ moniker in my opinion and hopefully this might be addressed by a firmware update down the line. Thankfully, there’s no such problem once DAW and audio applications are open and in focus in both mobile and desktop contexts; the audio stream stays awake once it’s up and running. During testing, a brand‑new set of batteries lasted around 90 minutes, with phantom power being drawn from one of the mic inputs for only part of this time. Prospective users would do best to invest in lots of rechargeable batteries for mobile use, and/or stick to using dynamic microphones — not unreasonable demands given the mobile capabilities of the interface. Despite some niggles, the iRig Pro Duo really packs punch as a truly mobile device, at a significantly lower price than its closest rival, the Apogee One (reviewed in March 2015’s SOS). Whilst it doesn’t have the same high‑quality preamp and converters as the Apogee, you get more audio ins and outs, MIDI I/O and Windows compatibility. Meanwhile, a few of the bonus software offerings help to seal the deal and will surely find a use in the home studio. Barry Watson $$ $199.99 WW

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Audio-Technica BP40 Dynamic Vocal Microphone If you’re looking for a mic that excels at speech intelligibility, Audio-Technica’s new broadcast model may be just what you’re after. HUGH ROBJOHNS


cursory examination of images of typical US commercial radio stations will reveal the near-ubiquity in that market of the Shure SM7B and the Electro-Voice RE20 for

broadcast presenter mics. So when I received Audio-Technica’s new BP40 dynamic mic — the company’s first dedicated broadcast voice microphone — I immediately thought that its designers had been inspired by the iconic shape of the RE20, too. There is a marked similarity in the visual design of these two mics, both featuring a large-diameter body with a big, square-ended, cylindrical mesh grille, with the mesh sitting behind a chunky frame of spaced rings. However, the RE20’s long body also incorporates a distinctive array of rear

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sound-entry ports (separated, again, by metal rings) for that model’s famous ‘Variable-D’ acoustic system, which is intended to mitigate the proximity effect. The BP40 lacks a comparable array, but its tough all-metal body is heavily machined with grooves of various widths, giving it something of the visual character of the RE20, at least. In fact, the widest of these grooves serves as the locating indentation for the supplied (AT8483) mounting clamp, and Audio-Technica state that these decorative channels were “inspired by the visual representation of a waveform” (although I might be rather worried to see such a flat-topped waveform at the output of a broadcast console!). The mic is 164mm in length, with a diameter of 56mm, and it weighs 632g. Its large, end-addressed 37mm moving-coil capsule has a hypercardioid polar pattern, and it’s mounted on a flexible internal suspension to provide some mechanical shock isolation. A humbucking coil placed directly above the diaphragm rejects interference from external magnetic fields, too. As this is a directional mic, sound needs to enter the rear of the capsule assembly as well as the front, of course, so the internal construction places the front diaphragm about halfway along the large dual-layer wire-mesh grille, roughly level with the front edge of the second metal ring. Positioned in front of the capsule, occupying almost the whole of the top portion of the grille, is a thick open-cell foam disc which provides a very effective pop screen. A standard male XLR is provided at the base of the mic body for the output signal, while the Audio-Technica name and model number are printed around the bottom

Audio-Technica BP40 $349 PROS

• Good protection from plosive popping. • Delivers a bright, crisp voice with good definition. • Attractive styling. CONS

• Optional shockmount rattles and appears too stiff. • Very strong presence boost may not suit all voices or applications. SUMMARY

Another cost-effective and well-engineered mic from Audio Technica, this is the company’s first to be aimed squarely at the broadcast presenter.

circumference of the mic body. I was expecting to find a serial number here as well, but there isn’t one, and it turns out that the serial number is actually found on a label on the internal support column directly below the capsule. What I did find, though, at the bottom of the mic body alongside the model number, is a miniature recessed slide switch that engages a high-pass filter. This has a gentle 6dB/octave slope from 100Hz, and is intended to compensate for proximity-effect bass boost rather than to serve as a rumble filter.

Shock & Roll Although the mic is supplied as standard in a padded, zipped leatherette bag, with a removable mounting ring-clamp, I was also provided with the optional AT8484 cat’s-cradle shockmount (pictured) for this review. The standard ring-mount locates in the largest groove near the base of the

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mic body, and attaches to a mic stand or angle-poise mount using a pivoting hub with a standard 5/8-inch thread (a 3/8-inch adaptor is included). Removing the clamp mount is not something you would want to do often, as it requires the complete — and I really do mean complete — disassembly of the mount, which takes some time and care! After removing the clamp I was left with the clamp ring, the hub, a wingnut, a metal washer, a coiled spring, a brass ferrule, two plastic washers and a bolt! The instructions supplied with the optional cat’s-cradle shockmount advise the reassembly of this removed mounting clamp “so no pieces are lost” — always a good idea! Fitting or removing the mic in or from the cat’s-cradle shockmount is, thankfully, a rather easier task. The all-plastic shockmount suspends an inner mounting carrier from two elastic loops zig-zagging from vertical arms on the inner ring to pegs on the outer ring. The pivoting bracket attached to this outer suspension ring has a 5/8-inch mounting thread as standard, but was, again, supplied with a 3/8-inch converter. To install the mic, a rotating section at the top of the inner mount is first turned counter-clockwise to release pressure on four captive, white plastic balls, allowing them to move outwards. The mics can then be lowered into place inside the mount, whereupon the top rotating section is turned clockwise, forcing the four balls inwards to locate in the wide groove of the mic body, effectively trapping the mic within the mount. The rotating ring latches positively into place to secure the mic safely, although the plastic balls do rattle very audibly if the mount is shaken or knocked,

Alternatives Cost wise, there isn’t much to choose between the mic reviewed here and the Rode Podcaster, or the Heil PR40, but the Electro-Voice RE20 and Shure SM7B cost about 20 percent more, while the EV RE320 is around 30 percent less.

which I found rather disturbing. The elastic suspension loops also seemed to support the mic rather more stiffly than I was expecting, and I suspect the shock isolation is not as effective at very low frequencies as it could be.

Specs Education Looking at the published specifications, the BP40 has a sensitivity of 3.9mV/Pa, which is about 6dB more than a Shure SM58, and the output impedance is unusually high at 450Ω — so would theoretically benefit from being partnered with preamps having an input impedance of 5kΩ or so. Although the marketing materials suggest the BP40 has a “rich, natural, condenser-like sound”, the frequency response is given as 50Hz-16kHz, with the range limits appearing to be the -10dB points (relative to the sensitivity at 1kHz, and with the source 12 inches or more away). Moving closer — which would represent a more typical use given its intended application — inevitably results in a low-end boost through the proximity effect, and the high-pass filter may then be required to restore a more natural balance. This filter introduces about 10dB of attenuation at 50Hz. Below 1kHz, the response is broadly flat and smooth down to 100Hz, albeit with a mild and broad lower-mid recession of a decibel or two. Above 1kHz, though, the

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response rises progressively to a prominent peak of +10dB at 4kHz, after which a series of gradually diminishing HF peaks return the sensitivity back unity by 15kHz. The high end rolls off steeply above that point, reaching -20dB at 20kHz. As a result, the BP40 has a strongly emphasised upper mid-range region, giving spoken voice a lot of clarity and presence — which is typically what is desired in a commercial style of radio broadcasting or podcasting. Most presenter broadcaster mics tend to have a relatively tight polar pattern, and the BP40 is no exception. The nominally hypercardioid pattern is well controlled across most of the frequency range, with stable rear nulls, and the pattern only really opens out around the frequency range of that high 4kHz presence peak. The rear tail is a good 15dB less sensitive than the on-axis response, and the deep nulls at 120 and 240 degrees offer at least 25dB of rejection: all very tidy and effective. Given Audio-Technica’s description of the BP40 as “condenser-like”, I reached for a classic moving-coil mic in my own collection to which I would give that exact same attribution: the AKG D224E ‘two-way’ microphone. This pinnacle of ’70s audio engineering employs separate high- and low-frequency capsules, their outputs being combined with a passive crossover much like a two-way loudspeaker in reverse. Its better-known sibling, the D202, was a firm favourite for broadcasters in the UK and Europe around that time, so it seems a very aposite reference comparator for the BP40.

diction, clarity and — that word again — presence! The peak is so strong that I can envisage some presenters’ voices not getting on with the BP40 at all. However, where there is a desire for a bright, crisp, clear sound — and especially if the signal chain includes a lot of compression processing — the BP40 delivers exactly what is required: a very modern, punchy and highly intelligible voice presentation. In comparison to other modern ‘broadcast announcer’ mics, the BP40 sounds much more similar to the Electro-Voice RE320 than that model’s more expensive sibling, the RE20. The former has a similarly strong presence peak, whereas the latter is much more linear, and more like the smooth-sounding AKG D224E. It seems the jury is out as to which is the more desirable or appropriate characteristic: the Rode Broadcaster mic takes a flat-response approach, while the Heil PR40 has a pronounced presence peak, and the Shure SM7B hedges its bets, as it can deliver either tonality with the flick of a presence-boost switch! The BP40 is certainly an effective mic for capturing voice, free from plosive popping and with a well-controlled polar pattern that will help to reject ambient studio noises. However, its conspicuous presence boost will inevitably divide opinion: some will love its strong, bright, crisp articulation, while others might prefer a more laid-back and neutral style — a decision that, I suspect, will also vary greatly with the broadcaster’s output genre and style.

Speaking Up Not surprisingly, the high presence peak of the BP40 really does make speech cut through, with quite a hyped sense of

$$ BP40 $349; AT8484 shockmount $99. TT Audio-Technica +1 330 686 2600 WW

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Artist Series / Artist EliteÂŽ Complete Live Sound Instrument Coverage With the addition of the Artist Series ATM230 and Artist Elite AE2300, Audio-Technica now offers a complete line of high-performance instrument microphones. Both models excel in high-SPL environments, but are designed for distinct applications. The hypercardioid ATM230 (available as a single mic or in a 3-pack) features a rugged, low-profile design and included drum mount, making it ideally suited for miking toms, snares and other percussion instruments. The cardioid AE2300, with its exceptional high-frequency and transient response, is perfect for miking guitar amps, brass and woodwinds, plus drums and percussion. A mic for every instrument. Line your stage with A-T.


Aston Microphones Halo Portable Vocal Booth

Could this British company’s new take on the portable vocal booth be the most effective PAUL WHITE


n the decade since the launch of sE Electronics’ original Reflexion Filter, the ‘portable vocal booth’ concept has grown so much that it now forms a market segment all of its own, and there are plenty of options from which to choose. The aim of all these products, of course, is to help avoid capturing unwanted room sound when recording. Regular readers might remember that, back in September 2014, we carried out a group test (http:// which demonstrated that there’s an inevitable trade-off: all such devices add at least some coloration. So if you’re recording in a great-sounding room, maybe you don’t need one. But, importantly, I find that the degree of coloration from

well-designed filters is vanishingly small, and in a typical home/project studio, it’s far less significant than the room sound they remove. You need to let your ears be the judge, but there’s no doubt that they can be very useful tools. One of my main concerns about the vast majority of existing solutions is that, as they’re open at the top and bottom, they do little to counter ceiling or floor reflections. Some products, such as the Kaotica Eyeball, provide more of an all-round screening solution, but with the ‘shield’ being very close to the microphone in that design, this comes at the expense of greater tonal coloration. All of this brings me neatly on

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to Aston Microphones’ Halo, the latest entrant to this market, because the Halo has been designed to tackle the floor/ceiling issue, while avoiding the heavier coloration of designs like the Eyeball — and, to my ears, they’ve succeeded. I’m told that Aston spent quite some time on research and development, evaluating the pros and cons of different existing products, exploring the properties of different materials and so on. For the absorption, they settled on a hi-tech acoustic PET (polyethylene terephthalate) felt. Seventy percent of this material is recycled plastic that’s subjected to a special process to form a porous, rigid felt. The overall shape is somewhat shell-like, wrapping around on all sides other than in the centre of the lower edge, where the

Aston Microphones Halo $299 PROS

• Controls floor and ceiling reflections. • Subjectively very little coloration — at least as uncoloured as anything else out there. • Lighter than similar quality competition. • Very stable on its mount. • About 25 percent larger than the sE Electronics Reflexion Filter. CONS





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The Halo may not be the first design to control floor and ceiling reflections, but it’s much more transparent than the others that do this. It delivers impressive results, and is also very easy to mount. July 2016 / w w w . s o u n d o n s o u n d . c o m

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support arm is fixed. Thanks to the PET felt, the Halo is very light in weight and this allows it also to be physically fairly large (54cm wide and almost 9cm thick at the centre). Thus, the mic is placed slightly further away from any of the reflective surfaces than on many similar designs — and that neatly gets around the coloration issue that I mentioned earlier, in relation to the Eyeball. The only downside is that this all means that the Halo a little bulkier than most and can’t be folded away; when not in use you’ll need space to store it. There are three layers of felt, which is a rather nice pale-purple colour. The front sheet and back sheet are the same, but the gap in between is filled with a lighter version. The ‘gap’ helps to improve handling of lower frequencies. The inside of the Halo is ribbed, so that any reflections that do come back from it are scattered to some extent. These ridges will only work at high frequencies, of course, but that’s all of any significance that’s likely to be reflected back, and even then only a little. The stainless-steel mounting hardware is engineered in the UK, and includes a cable tidy. Simple stand mounting is accommodated via a US thread, and there’s plenty of flexibility in setting the mic position, both in distance from the screen and in height. It is also surprisingly well balanced, sitting over the centre of gravity of the mic stand, so far less likely to topple than some products we’ve looked at, the original sE Reflexion Filter being one (though this was addressed in some later variants).

Tests I tested the Halo for a vocal recording session (there’s no reason it can’t be used

for other sources) and found it easy to set up and easy to adjust. It proved very effective in screening out room ‘live-ness’ but, as with all other such devices, if you are working in a seriously bad room, then you should also put a further large absorber behind the vocalist — a foam panel or a polyester duvet would do the job. This is to intercept reflections from the wall behind that might otherwise find their way into the open side of the screen, and onto the hot side of the mic. My own studio is fitted with basic acoustic treatment, so I used the Halo on its own. The subjective results were perfectly clean; I could discern no audible coloration. Though I was acquainted with the specs beforehand, I was still surprised when setting it up at just how rigid and lightweight the structure is. I found it to be perfectly stable when placed on a dedicated bog-standard mic stand (ie. on a different stand from the mic), and the extra isolation this produces is definitely worth it.

Conclusion I’ve been impressed already by Aston’s microphones, and the first of their accessories is no different. In fact, I think the Halo sets a new bar for personal vocal booths: the large, lightweight shell-shaped design and the felt material combine to make it very effective. Sure, its size and shape mean it isn’t the easiest thing to store if you’re in a small room, but that’s a very small price to pay for something that works so well. Highly recommended. $$ $299 TT Presidio Label +1 805 895 9919 WW WW

July 2016 / w w w . s o u n d o n s o u n d . c o m


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Sonic Farm Silk Road Microphone Preamplifier

The latest preamp from Canadian company Sonic Farm is a classy and versatile affair, thanks to its wide range of tonal options.



’ve had a few Sonic Farm products across my test bench in recent years, including the Creamer (SOS May 2013) and Berliner (SOS May 2016) valve mic preamps. Sadly, I didn’t get to try the Silkworm 500-series solid-state mic preamp module, as that pleasure instead befell my colleague Bob Thomas. However, it seems he was very impressed with it (SOS October 2014), and that’s good to know because the Canadian device that’s sat beside me now — the

Silk Road dual-channel desktop preamp — is quite clearly derived from the same design. So much so, in fact, that clicking on the ‘download manual’ button on Sonic Farm’s web site delivers the Silkworm’s manual! It’s not unusual for manufacturers to ‘re-house’ 500-series modules as desktop or rackmount products. After all, why go to all the trouble of redesigning PCBs when often all that’s needed is the addition of a power supply? However, that’s not quite what Sonic Farm have done here. The Silk Road clearly employs

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bespoke new circuit boards, as well as a slightly different control layout and the welcome addition of a configurable high-pass filter. At its core, though, this is the same high-quality solid-state preamp, complete with its fully discrete gain stage constructed from carefully hand-matched transistors, and with a DC-coupled signal path to avoid any possible capacitor distortion and unwanted phase shifts. Of course, the downside of DC-coupling is a risk of internal DC offsets (which can cause asymmetrical clipping) and nasty pops and bangs when switching the output signal (or editing the resulting audio in a DAW). However, a dual-stage DC servo system takes care of all that, and the end result is a preamp which enjoys a fundamentally clean, neutral, and fast character, with extremely low distortion and massive headroom. Having said that, in the traditional Sonic Farm way, this preamp still has plenty of options for introducing some musical colour too, if required.

Sonic Farm Silk Road $1800 CAD PROS

• Fundamentally fast, clean and quiet sound character. • Vibe mode and selectable output transformer allow a wide range of sonic tailoring. • Useful high-pass filter options. • Musical instrument input mode. • Integrated universal power supply. • Excellent technical specifications and build quality. • Handy dual-channel package. CONS

• The on-off position of the toggle switches is unclear. SUMMARY

Road Trip

A dual-channel re-packaged and extended version of the Silkworm 500-series module, featuring a discrete transistor gain stage, interesting input impedance variations, and a high-pass filter, along with Sonic Farm’s classic feature of selectable transformer or solid-state output configurations.

The Silk Road is constructed as a self-contained desktop unit, housed in a bright-red, brick-shaped steel case with a flexible retracting carry-handle on the top. If you’re eyeing a convenient space at the side of your desk you’ll want to know that it measures 150 x 110 x 305mm (WHD) and weighs 2.9kg. The rear panel contains four XLRs for the two sets of mic inputs and line outputs, along with the usual IEC mains inlet and fuse holder, plus two small rocker switches for mains on/off and a ground-lift facility. The internal switched-mode power supply module accepts 100-240 Volts AC and

generates ±24V audio power rails — and since these are rather higher than those of the Silkworm 500-series module, the Silk Road desktop preamp enjoys a greater headroom margin and higher maximum output level (all the way to a massive +32dBu). Moving to the front panel reveals a veritable forest of miniature toggle switches, each channel bearing no fewer than nine. There’s also a rotary gain trim control operated with a white vintage-style knob. The two channels’ control sets are clearly separated one

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above the other, and it appears that the ‘off’ position for the toggles is towards the right (something the labelling fails to make clear!). Each channel features a quarter-inch unbalanced instrument input socket at the extreme right-hand side (with a 1MΩ input impedance), while to the extreme left of each channel is a pair of toggle switches to configure a 12dB/octave high-pass filter. The upper, three-position toggle selects turn‑over frequencies of 80, 160, or 320 Hz, while the lower toggle turns the high-pass filter on or off. This facility does not feature at all on the Silkworm 500-series module (presumably due to panel space restrictions), and it’s extremely useful. The published specifications align to my own Audio

Precision measurements, the plots for which can be found on the SOS web site at: All of the remaining controls are identical to those of the Silkworm, even if some are in different positions. The more familiar functions include an input selector for the rear-panel mic or front-panel instrument connections, a 15dB input pad (pre-transformer), output polarity reversal, and a soft-start 48V phantom power (with red LED indicator). The phantom power voltage measured comfortably within specifications even when delivering the maximum current. An adjacent bi-colour LED provides basic level metering, showing green with variable brightness for a healthy output signal, changing to red at +29dBu as clipping approaches.

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Anyone familiar with other Sonic Farm products will recognise the ‘OT/ SS’ switch, which allows the user to route the output signal through either a Cinemag output transformer (with 100‑percent iron core) or via a standard solid-state balanced output driver chip. As you’d expect, the transformer output sounds a little rounder and fatter, thanks largely to that iron core, and this effect is sensitive to the output drive level. The solid-state output is, in contrast, very clean and neutral at all levels. Coarse gain is set with a three-position toggle switch (labelled ‘H, L, M’) in roughly 18dB steps, although it’s worth

“The Silk Road preamp is, as I’ve come to expect of Sonic Farm, very well-designed and constructed and performs to exemplary standards.” noting that the ‘H’ and ‘M’ modes provide the same gain range for instrument inputs (because the high-gain mode is not appropriate in that case). The conductive-plastic rotary control provides a continuous trim for the selected gain mode over roughly the same range. In this way any required gain can be achieved easily for the mic input between



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+3 and +67 dB. For the technically minded, the precise mic input gain ranges are 53.1 to 67.6 dB (H), 37.5 to 52.8 dB (M), and 18.3 to 38.5 dB (L), with the pad switch reducing everything by a further 15dB. In practical terms, the maximum mic input level to generate a +4dBu output is +2dBu with the pad engaged, but as clipping doesn’t occur until +32dBu the absolute maximum input level is a whopping +30dBu. The minimum input level necessary to generate a +4dBu output is -63dBu. For the instrument input, the H and M settings share the same gain range of 26.9 to 42.1 dB, while the L mode offers 7.6 to 27.8 dB, and the pad is not active on the instrument input. The maximum instrument level for a +4dBu output

is -4dBu (gain set to L), and the minimum is -38dBu (with gain set to M or H). Again, given the massive headroom margin, the absolute maximum input level is +24dBu, which is more than enough for any electric instrument. An unusual feature, borrowed from the Silkworm, is the Vibe switch, which was originally going to be labelled ‘character’ — apparently there wasn’t space on the panel for that so it was rechristened Vibe! It uses capacitor-resistor (CR) networks to alter the impedance of the microphone input in a frequency-selective way. It has three modes, identified as Smooth, Present and Warped (S, P, and W, on the switch), where the ‘P’ mode provides a nominally flat response. In contrast, the ‘S’ mode rolls off the extreme highs

July 2016 / w w w . s o u n d o n s o u n d . c o m


(giving the impression of warmer lows and mids), while the ‘W’ mode adds some extra ‘air’ around 8-10 kHz. As is always the case with variable input-impedance designs, the effects are most apparent with dynamic (moving-coil and ribbon) microphones, and least audible with capacitor mics or those with active outputs — although the inherent frequency response characteristics of the CR networks will exhibit a much more apparent effect with capacitor mics than more conventional variable-impedance

preamp designs would. As well as altering the frequency response, these CR networks also affect the phase response: ‘W’ mode introduces considerable phase lag at high frequencies in comparison to ‘P’ mode, while ‘S’ mode (green) brings in a noticeable phase lag in the mid band. Engaging the output transformer while in the default ‘P’ mode (orange) also brings in some modest phase lead at LF and lag at HF, of course. I tested the input impedance (with

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phantom power off) for the various Vibe modes using an NTI Minirator and, given the reactive circuits involved, I tested at 100Hz, 1kHz and 10kHz. The default ‘P’ mode presented an input impedance of 3.5kΩ at 100Hz rising to 5.2kΩ at mid and high frequencies. So it’s not quite the 8kΩ mentioned in the handbook, but still usefully higher than most stock mic preamps. Switching to ‘S’ mode, the impedance remained at 3.9kΩ at 100Hz but fell dramatically to 760Ω at 1kHz and 560Ω at 10kHz. The ‘W’ mode values were 3.6kΩ at 100Hz, 4.1k at 1kHz, falling sharply to 470Ω at 10kHz. These radically different input impedances at different frequencies alter the loading on a dynamic microphone’s capsule, and thus affect its frequency response, and that effect is compounded by the frequency dependent responses of the CR input networks themselves. Not surprisingly, the pad switch reduces the input impedance for all Vibe settings,

Alternatives Surprisingly, I can’t think of many dual-channel solid-state preamps with instrument inputs in a desktop-format, let alone any with discrete transistor gain stages or the tonal versatility of the Silk Road!

and engaging phantom power brings the input impedance back to around 2.2kΩ for every mode (because of the extra phantom feed resistor loading). Checking the distortion measurements, the Silk Road produced 0.01 percent THD when generating an output level of +24dBu (at 1kHz), regardless of which output mode was selected. Reducing the test frequency to 100Hz revealed a THD+N figure of 0.05 percent for the solid-state output and a much higher 0.26 percent for the transformer, which is entirely as expected... and desired!

In Use The Silk Road preamp is, as I’ve come to

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expect of Sonic Farm, very well-designed and constructed and performs to exemplary standards. Whereas most of the other SF products are clearly designed to impart some musical colour onto the microphone’s signal, the Silk Road is fundamentally a much more neutral device, although the ability to select an iron transformer for the output allows some vintage character to be dialled in, and that is supplemented by the ‘S’ and ‘W’ Vibe modes which tailor the signal even more strongly. So in reality, the Silk Road offers six distinctly different sound flavours — many of which will also vary interactively with your dynamic microphone collection! Whereas the valve-based Sonic Farm preamps I’ve reviewed rely heavily on various flavours of harmonic distortion for their sonic interest, the Silk Road is much more about frequency sculpting, which opens alternative doors to sonic creativity. There’s no doubt at all that the Silk Road dual preamp is an unusually versatile and very high-quality preamplifier, exhibiting all the qualities we normally associate with

discrete transistor designs. It’s fast and clean, quite API-like in many ways, yet the use of Cinemag input and (selectable) output transformers, combined with the switchable Vibe tonalities provide everything from clean, snappy and modern sounds to coloured, smooth and distinctly ‘vintage’. The instrument input is also very clean and quiet, but certainly not bland or sterile in any way. The only niggle I would raise — but one which may well not apply to other potential customers — concerns the aesthetics of its forest of toggle switches and their ill-defined on/off positions. Nevertheless, I enjoyed using the Silk Road very much indeed. In fact, of all the Sonic Farm preamps I’ve tested to date this is probably my favourite and the one I’d be most likely to buy myself, as it’s ideally suited to the kind of classical and acoustic work I prefer. $$ $1800 CAD (about $1400 USD). TT Sonic Farm +1 310 402 2390. WW

July 2016 / w w w . s o u n d o n s o u n d . c o m



Greg Wells Plug-ins Mix Bus, Piano & Voice Processing Plug-ins For Mac OS & Windows July 2016 / w w w . s o u n d o n s o u n d . c o m


A trio of easy-to-use plug-ins from Waves aims to bottle the signature sounds of pop producer Greg Wells. PAUL WHITE


vailable as separate products, but also as a suite, Waves’ latest series of signature processors is designed to fulfil specific mix applications where ease and speed of use are important. MixCentric is designed to be used as a bus or overall mix processor, while KeyboardCentric is suited to refining piano and other keyboard sounds, and following the same logic, you can probably guess what VoiceCentric is intended for! If the name Greg Wells is not familiar, suffice it to say that he’s a very successful producer, songwriter, musician and mixing engineer who’s been nominated for a Grammy award and has worked with artists such as Adele, Katy Perry and OneRepublic. To create this series of plug-ins, he worked closely with Waves to emulate his own typical processing chain for each task. All three plug-ins are available in mono and stereo versions for all the common Windows and Mac OS formats, in line with the rest of the Waves range. They can be authorised to a specific computer, to the Waves Cloud or to a standard USB stick. The plug-in header bar is also consistent with the rest of the Waves line, with preset loading and saving functionality, A/B switching between settings and so on all available as usual.

MixCentricity MixCentric isn’t a complete, one-stop mastering plug-in in the vein of iZotope’s Ozone or IK Multimedia’s T-Racks; it does

not, for example, include a brickwall limiter, but its focus instead is the specific processes a mastering engineer might choose to employ in trying to add density and life to a mix or submix. With all of these plug-ins, setting the input and output levels correctly is important, so there are both input and output level sliders, each with a long meter. The input needs to be set just high enough to have the indicator to the left of the large control knob hover around the yellow zone — green is a touch low and red means you’re in danger of overcooking the process. Once this is set, the output fader can be adjusted so that you hear the same level when the plug-in

Waves Greg Wells Plug-ins $349 PROS

• Extremely easy to use. • Can produce sophisticated effects that would usually require several separate plug-ins. CONS

• The preset nature of the processes might feel a little restrictive to more advanced users. SUMMARY

All three Greg Wells plug-ins are capable of excellent results on a wide range of material — though it still pays to learn how to create effects the ‘hard’ way, for when you come across something that requires an unusual treatment!

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is bypassed as when it is active. This saves you falling into the ‘loudest sounds best’ trap. While MixCentric looks simple, and indeed is simplicity itself to operate, there’s a lot going on behind the scenes. Once the levels have been set, the large central Intensity knob is basically a ‘more of everything’ control. It starts out doing relatively little, and then piles on increasingly more processing as you move further clockwise. A gain-reduction meter shows how much compression is being added, and this can be fine-tuned

by adjusting the input level fader setting. Once you get past halfway around the dial the sound becomes more cohesive and starts to take on an up-front character. Keep going and you can dial in just a little aggression, which could work well for livening up a drum bus mix. If you have the expertise, you could achieve similar results by combining EQ, compression, harmonic enhancer-style enhancement and subtle overdrive, but this plug-in really takes the effort out of doing it the hard way. While it is perfect for those who lack confidence

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with complex combinations of plug-ins, the results are good enough that I’m sure it will also prove be attractive to professionals who just want to save a bit of time.

The Old Joanna The PianoCentric plug-in follows a similar format, but with two extra controls, which add variable amounts of preset delay and doubling treatments. This time the central control, which has keyboard-style markings around the edge, has a centre-neutral configuration; moving anti-clockwise produces a progressively darker and more lo-fi sound, with quite peaky filtering that lops off both the highs

and lows, while moving clockwise from centre brightens the sound in a way that makes it stand out in the mix. Some of the filtering on the ‘dark side’ sounds a bit extreme in isolation, but within a mix it can work spectacularly well. I suspect there’s more going on here than just EQ, though the exact recipe remains a secret. The doubler probably owes something to the Waves Real ADT and sounds very organic, unlike regularly pulsating chorus effects. Before trying this plug-in I wondered if there was a need for it — a piano is a piano, right? — but having tried it I can confirm that in the world of pop, where piano sounds sometimes need to be ‘bent to fit’, it is a wonderful tool.

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It is remarkable how a fairly boring piano sample can be transformed into something either lush or assertive, and that extreme filtering pays dividends in shoehorning a piano into a tricky mix without allowing it to take up too much space. The preset delay works for most things if mixed in at suitably low level, while the doubler is just magical in creating fat, lush pop or ambient piano sounds. I tried a few experiments with synths and other sampled sounds, too. It won’t work for all synth sounds but it can dish up a few alternative flavours, some of which may surprise you.

Self Centred I don’t know how many times we’ve been asked how to make a vocal sound more ‘produced’, but it must be rather a lot. Usually it’s about a tasteful blend of EQ, compression, reverb, delay and maybe some other types of ‘fairy dust’, but the new Greg Wells VoiceCentric Plug-in is about as close as it gets to a one-knob solution. Again, the user interface is almost startlingly simple; in essence, there’s one large Intensity knob that controls the strength of the main ‘produced’ effect and three smaller knobs that adjust the amount of added Delay, Doubler and Reverb. As with PianoCentric, these effects are pretty much preset, so the only control you have is over level. Intensity adjusts multiple parameters to increase the subjective strength of the underlying processing, with more extreme settings sounding brighter and more ‘in your face’. Once you’ve arrived at a setting that flatters the timbre of the voice you can then go on to adjust the three effect level controls. Those effects

are tastefully tuned to produce a polished result with minimum effort; the delay seems to have been equalised to make it sit nicely behind the vocal, while the reverb just adds the required amount of ‘wet’ without seeming to get in the way. Again the doubler seems similar to the Waves Real ADT plug-in and manages to create a plausible double-tracked sound by adding delay and also varying degrees of pitch/time change. At lower settings it thickens vocals without being too obvious, whereas at maximum you get something akin to the full John Lennon treatment. I tried this plug-in on a number of tracks I’d already mixed, in place of the three or four plug-ins originally deployed on those tracks, and I have to say the results were very encouraging. The processing pushes the vocal nicely to the fore and the choice of effects is very supportive, while if you want more reverb, you can easily add your own using a separate plug-in. I preferred settings around the middle of the Intensity dial but if you’re into more assertive vocal styles, the upper reaches also produce artistically useful results. Summing up then, these may be ‘instant gratification’ plug-ins, but they are far from being mere toys. I very much enjoyed working with them, and they really do produce polished results with minimal effort. They’re easy to use, affordable and effective. What’s not to like? $$ MixCentric $199; PianoCentric and

VoiceCentric $149 each; Signature Series bundle $349. TT Waves +1 865 909 9200 EE WW

July 2016 / w w w . s o u n d o n s o u n d . c o m



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Bright Sparks Film & Album


he history of synthesis has already been written many times, and numerous books (some well researched but others much less so) purporting to tell the stories and evaluate the instruments that adorn them have been published over the past 30 years or so. Today, the situation has improved enormously; I recently reviewed a couple of books that proved to be both meticulously researched and cracking good reads. Now it’s time for the video world to catch up. There are a handful of gems out there, but many of the broadcast documentaries lack understanding of the subject matter and perpetuate every irritating stereotype the writers could employ. Clearly, there’s a need for an oral history told by the people who were there at the time, and who are able to reminisce about the people, the companies, the arguments and the decisions that lay behind the published histories. Strangely, Bright Sparks was originally conceived by Jarrod Gosling and Dean Honer of the band I Monster as an album that honoured specific synthesizers (or families of synths), by featuring them on individual tracks with lyrics telling their stories. They decided to focus on eight that they regarded as seminal: Moogs, Buchlas, ARPs, the Chamberlin and the Mellotron, the EMS VCS3, the EDP Wasp, and the Freeman String Symphonizer. But as they didn’t have all of the equipment necessary to make this a reality, they approached Dave Spiers and Chris Macleod of GForce to ask whether they

could borrow some of the instruments that they needed — the chaps at GForce were happy to assist. One day, out of the blue, Honer asked Spiers whether he would be interested in making a short movie to accompany the album. Spiers agreed because, for reasons we need not discuss here, he’d realised that the time would come when the original pioneers were no longer with us, and the opportunity to hear their first-hand accounts would be lost. Not wanting to reduce the proposed interviews to soundbites, he suggested that, rather than make a promo video, they should attempt a cinematic-length documentary. Everyone agreed, so Spiers and Macleod spent a year travelling, interviewing and editing, often coming away from the sessions with fascinating accounts that illuminated the accepted histories. When filming was complete, the

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collaborators decided they’d release the album and the documentary simultaneously; a world premiere was screened in Reading, UK, late last year. Like the album, the movie is split into eight chapters, opening with an account of the birth of Moog synthesizers. This is fascinating because, when Herb Deutsch (who started working with Bob Moog in 1964) discusses their adoption of voltage control, the invention of the 1V/octave tuning standard, and the arguments over whether a synth should have a keyboard or not, he’s not discussing received history, he’s telling you about the actual discussions he and Bob had more than

50 years ago! Others chapters include: Alan Pearlman, who for decades had declined to be placed in the spotlight but here talks freely about the initially prickly relationship that he had with Moog; ARP engineer Dennis Colin who co-designed the ARP 2600; John Bradley of Streetly Electronics, who is much more forthright than most about the incidents leading to the birth of the Mellotron; a wonderfully candid Peter Zinovieff of EMS; Fred Gardner of EDP; and Ken Freeman explaining how he conceived the idea of the string synth. The result is a documentary that isn’t merely well informed, it’s definitive. And, with no

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silly camera angles, no stupid lighting, no intrusive background muzak, and no trendy (and unwatchable) shaky cameras, it’s a pleasure to watch, almost as if the interviewees are sitting in your own living room or studio. All of the music for the video and album was written and played by I Monster, with just a handful of additional performers. At this point, I have to declare an interest, because I played the Freeman String Symphonizer on the wonderfully titled, ‘The Further Adventures Of Ken Freeman & His Incredible Machine Of A Thousand Strings’. But all of the tracks deserve repeated listening because, as well as being damn good music, they often use the instruments in ways you may not

expect. And how can you resist tracks called ‘The Wizards Of Putney Deny Accusations Of Unholy Enchantment At The Electronic Music Studios’ and ‘Alan R Pearlman & The ARPiological Exploration Of The Cosmos’? Bright Sparks isn’t just another tribute to analogue synths; it’s perhaps the most successful project yet to persuade the sometimes shy and always modest pioneers of the ‘60s and ‘70s to tell the human stories behind the electronics. As such, it’s warmly recommended to everyone who has an interest. Gordon Reid $$ Film £8. Film & album £15. WW WW

Hamstead Soundworks Signature Analogue Tremolo Guitar Effects Pedal


hen I reviewed Hamstead Soundworks’ Artist 20+RT combo in SOS November 2015 I gave high praise to the sound of its solid-state analogue tremolo. Unlike the bias-modulating tube circuitry in my vintage Deluxe Reverb, the Artist 20’s tremolo produced a beautifully smooth cycle with no extraneous noise. Fast‑forward a few months, and designer Peter Hamstead has had the novel idea of combining the Artist 20’s tremolo with an independent variable 0-10 dB clean gain boost, to produce that which I’m tempted to describe as an effects pedal with a little bit of an attitude. In addition to the Artist 20+RT tremolo’s ‘Signature’ sound, the Hamstead pedal can be switched to a ‘Classic’ mode,

mimicking the harder-edged lumpiness of a vintage amp’s tremolo section. Controls and switching for both Tremolo and Boost functions are simple and intuitive, although personally I would like to see some scale marking around the knobs.

“The Hamstead Soundworks Signature analogue tremolo pedal, like the company’s amps, is superbly engineered and sounds wonderful.” Being too power-hungry to run off a 9V battery, the Signature pedal requires an external 9-18 V DC power

July 2016 / w w w . s o u n d o n s o u n d . c o m


supply that’s capable of supplying at least 300mA of current — this must be purchased separately if you don’t already possess one. An internal DIP switch gives you the option of running the Signature in either true-bypass or buffered modes. Despite the fact that, in true bypass, all the pedal’s active circuitry is out of circuit, I could hear no tonal difference between the two, the only real difference being that there is no switching noise to be heard when in the buffered mode. Since that mode is the one you’d be using if the Signature was driving other pedals down the line and there’s no sonic penalty, I’d tend to leave it there. Having made your choice, in operation the tremolo’s Signature setting sounds exactly as I remember it from the Artist 20+RT — smooth and musical across the entire frequency range. Classic mode sounds to me more old-school and somewhat rougher-edged in comparison but is, nevertheless, more than capable of putting sounds from the ’60s and the early ’70s at your disposal. A goodly amount of clean gain is always useful, not only to drive the input of a valve amplifier that bit harder, but also to beef up single-coil pickups. Placing a boost pedal in an amplifier’s effects loop, rather than over-pushing its preamp into distortion, is often well worth trying if you want a bit more level. With the Signature pedal, you can use the boost with or without the tremolo (or the other way around) so, if you insert it into a non-tremolo amp’s effects loop, you can add both tremolo and boost post preamp — especially if you have another boost pedal that you can put in front

of the amp. The Hamstead Soundworks Signature Analogue Tremolo pedal, like the company’s amps, is superbly engineered and sounds wonderful. There are quite a few competitors with similar products, both analogue and digital, sitting around the same price point but, from what I can see, the Signature’s particular combination of tremolo and independent clean boost seems to be unique. I could perhaps quibble about the absence of an included power supply, but in reality I can’t think of a reason why anyone in search of a great-sounding high-end tremolo pedal would not consider buying a Hamstead Signature. If you’re looking for a tremolo, you should try out a Signature — you’ll like what you hear! Bob Thomas $$ £165 (about $240). WW

July 2016 / w w w . s o u n d o n s o u n d . c o m


Electro-Harmonix Mel9

Mellotron Emulator For Guitar


ollowing on from their B9, C9 and Key9 pedals comes Electro-Harmonix’s Mel9, which must qualify as the most ambitious of the bunch. It has the same case and control layout as the other pedals and works in a similar way. As far as I can tell, in all these ‘transformative’ EHX pedals, the sounds always start life as waveforms taken from the guitar strings, but some clever polyphonic pitch-tracking filters combine with pitch-shifting to allow both single notes and chords to be processed in real time to make them sound, in the case of the B9 and C9, like electronic organs or, in the case of the Key9, like electric pianos. The Mel9, of course, takes on a range of instantly recognisable Mellotron sounds. The Attack control

governs the rate of attack of the note and Sustain allows it to linger on for a short while after you end the note, to smoothen chord changes. The only deviation from this is when playing the Brass patch, in which case the Attack control adjusts a dynamic filter and Sustain imparts a ‘parpy’ lip noise to the sound. As with the other pedals in the series, there are separate outputs for the dry and effected sounds, though plugging into only the effect output gives a mix of both wet and dry sounds, each with its own level control. You can use the pedal right out of the box but it needs a strong signal (which means keeping the guitar volume turned up full) to keep it happy, and it’s extremely sensitive to picking dynamics, so unless you are a very precise player, the note levels can seem very uneven. I found this to some extent with the other pedals in the series, and discovered a simple fix: just put a compressor pedal in front of it. I had an EHX Soul Preacher lying around and it did the job perfectly — which makes me wonder why they didn’t incorporate compression in the pedal. It might hike the price slightly but it would be worth it in my view; it might even help them sell more, as it would make the pedals more user-friendly when being tried out in a music store! The Sustain control works well up to around halfway on the control, after which it starts to sound very ‘reverby’ so I’m guessing some sort of reverb engine is used to create the sustain. The sounds on offer are Orchestra,

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Cello, Strings, Flute, Clarinet, Saxophone, Brass, Low Choir and High Choir, of which more shortly. Orchestral reproduces that characteristic ‘Hall Of The Crimson King’ sound with almost frightening accuracy, to the point you can almost hear the tapes shedding oxide and stretching. Next is Cello, which sounds convincing enough, then there’s a generic String sound that again works fine. Flute delivers the ‘Strawberry Fields’/’Nights In White Satin’ tonality, while the Clarinet and Sax are, just like the originals, recognisable but not up there with today’s multi-samples; old-world charm is most definitely the aim here. Brass conjures up a believable ‘Bus Fare For The Common Man’ sound, and can also be softened to work as

a pad, while the two Choirs are again very close to the originals, complete with warbly vibrato. If you need these sounds and don’t want to mess around with special guitars or pickups, the Mel9 will fool most people into thinking they’re hearing a real Mellotron when used in a live performance. But do consider adding a simple compressor to the front of the chain, as that really improves the evenness of the sound and the playability. It may offer stompbox simplicity of operation but what the EHX engineers have achieved here is quite extraordinary. Paul White $$ $221.30. WW

Samson Z55

Closed-back Headphones


hen you think of headphones, Samson might not be the first name that springs to mind, but their top‑of‑the‑range Z55 Professional Reference Headphones are certainly worth a listen if you’re looking for closed-back studio headphones that aren’t hyped to ‘enhance your musical experience’. These ‘phones are reasonably light in weight (240g) and designed to provide a practical amount of sound isolation if used for tracking. The collapsible rotating earcups allow the headphones to be stored flat, and a soft carry pouch is included. A welcome feature is the detachable cable or, to be more precise, three detachable audio cables, which give the user the option of straight or

coiled cables, or a cable with a built-in microphone and call/answer control button for phone or Skype calls. All three cables terminate in 3.5mm mini jacks but an adaptor for quarter-inch outputs is included. The headphone end of the cable is also a 3.5mm stereo jack but with a twist-to-lock fitting. If you plan to use any headphones in the studio for any length of time they need to be comfortable. To this end Samson have made the ear pads from lamb skin, rather than synthetic leather. These are ‘on-ear’ type phones, which means the pads sit over the top of your ears rather than around them. A cushioned headband allows adjustment to any head size. All in all they’re very comfortable. The transducer is obviously a vital

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component, and Samson have chosen a 32Ω 45mm driver, built with a copper-clad aluminium voice-coil capable of a 10Hz-25kHz frequency response. These are powered by ‘rare earth’ magnets, and have a quoted sensitivity of 98dB at 1kHz, at their maximum 1000mW power handling. There’s more than enough volume available for any purpose I can envisage, even driven straight from an iMac headphones output, and the overall tonal balance is clean and detailed, though just a fraction more ‘lively’ than my Beyer DT770s. There’s a good sense of space to stereo material, clear detail, and lows that extend down a long way below what you might expect from typical studio monitors

without becoming boomy or hyped. I’d urge you to try the Z55s alongside any other closed ’phones you might be considering, as what constitutes the ideal tonality is open to personal interpretation. Although I’d normally suggest open phones for mixing, these would work fine as dual-function tracking and mixing phones if you can’t run to two pairs. Being best known for their budget products, Samson might find it difficult to persuade users to put their ‘phones up against comparably-priced models from the likes of AKG, Beyerdynamic, Sony, Audio-Technica and Sennheiser, but the reality is that the Z55s deserve to be taken seriously. Paul White $$ $199.99. WW

Mojotone Quiet Coil

Noiseless Guitar Pickups


oiseless passive single-coil‑sized pickup designs have progressed enormously, from the earliest ‘stack’ models to the more complex current designs from the likes of Kinman, DiMarzio and Fender. For players who need to keep induced noise (hum) at bay whilst still using Fender-style single-coil pickup sounds, many of the current products are now far more than just the ‘acceptable compromise’ that they used to be. Mojotone’s Quiet Coil

design enters the noiseless ‘single-coil’ market with two models, the ‘58, offering the tonality of a late ’50s Fender Strat pickup, and the ’67, replicating the tone of the Strat pickups of the late ‘60s. Unlike the majority of noiseless single-coil designs, these utilise side-by-side coils, rather than the commonly favoured stacked-coil configuration — effectively, they are a conventional humbucking design. Of course, single-coil‑sized

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side-by-side humbuckers have existed for years without anyone claiming that they sounded all that much like a true single-coil pickup, so what’s different here? Quiet Coil designer David Shepherd has taken the very simple approach of using the same 42-gauge Heavy Formvar wire and Alnico magnets as a real vintage Strat pickup and configuring them as two tall, narrow coils, each with half the normal DC resistance. A series connection between the coils restores the combined DC resistance to typical single-coil values (5.8kΩ in the ‘58 model and 5.6kΩ in the ‘67 model), while a reversed winding and reversed magnetic polarity in one of the

coils creates the hum-cancelling effect. The visible ‘pole pieces’, complete with ‘high-G’ vintage magnet stagger, actually don’t do anything at all; they are purely to maintain a conventional appearance. Some people may balk at this being purely cosmetic inclusion with no actual function at all but I must admit, I rather like it; Strats somehow don’t look ‘right’ without conventional pole pieces to me. I tested a ‘58 Quiet Coil set — 5.8kΩ neck and middle pickups with a 6.2kΩ bridge unit — mounted on a Mojotone pre-assembled pickguard, using the recommended 500kΩ volume pot (don’t be tempted to try to make do with your existing 250kΩ pot:


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500kΩ does seem to be necessary to get the proper voicing of the pickup). Tonally, I found them to be remarkably Strat-like: crisp and articulate, but never harsh, and without the compressed dynamics that many stack designs exhibit. When you’ve experienced a lot of different pickup designs, you know when the resonant peak — the sonic signature of the pickup — is in the right place, and the Quiet Coils have very much got that right. True single-coil pickups are all about detail — the little scrapes and ringing noises that all come through loud and clear — which is what makes them both so satisfying and expressive, but also very revealing at the same time. What is so impressive about the Quiet Coils is that, up close and personal, with a clean tube amp in a quiet environment, so much of that ‘real single-coil’ detail is still there. Unsurprisingly, a really nice vintage, true single-coil still beats it for overall complexity, but by the time you are at performance levels, with some distortion involved as well, I think player and audience alike would be hard-pushed to tell the difference any more. The Quiet Coils are very good at resisting induced noise picked up by the

coils — they are actually better than most full-size side-by-side humbuckers, due to the tighter proximity of the coils — but you’ll only keep out other sources of noises with a good overall screening job. I tested the Quiet Coils in a well-screened guitar and could generate no hum or buzz under any normal performance conditions. The manufacturer’s notes suggest you can adjust the Quiet Coils closer to the strings than a normal single-coil without side-effects, partly because of the lower Gauss magnets, but also because you have both polarities in close proximity, effectively both pushing and pulling the strings at the same time, rather than just pulling. There’s no dramatic tonal change when you get them up close, unlike Kinmans which offer a lot of tonal tuning by proximity, and the output level always remains comparable to a conventional single‑coil, too. ‘Quieting’ a single-coil guitar has always been about choosing your compromise. Most of mine now have Illitch plates, and I just accept that there will be one angle at which they may still hum (generally solved by not standing at that angle...), and I keep a Strat with early Kinmans on it precisely because I love their squishy vintage ‘softness’ — the very characteristic that causes some people to look elsewhere. There is no ‘magic’ new design or technology here: it really is all down to the detail of the construction and material choices, but Mojotone’s Quiet Coil models are right up there with the best of the alternatives and offer a convincing single-coil tone and dynamic response with absolutely no hum. Dave Lockwood $$ Both versions $229.95 per set or $299.95

for a loaded scratch plate. WW July 2016 / w w w . s o u n d o n s o u n d . c o m




The new MX Series powered studio monitors extend Sterling Audio’s unparalleled reputation in advanced transducer technology. Each is designed and built with the same commitment to quality that has made Sterling studio microphones the top choice for producers and engineers the world over. Combining superior sound transparency with next-generation materials, these reference monitors offer high efficiency and ultra-low distortion in 8” and 5” woofer configurations. The Sterling MX Series is the ideal audio solution for any value-conscious studio.

1” silk dome tweeter utilizing neodymium magnets give the MX Series monitors a smooth and natural high-frequency response.

Sterling’s dual-axis WaveGuidanceVH™ technology creates an incredibly wideand-high audio “sweet spot”, giving you clear, articulate and centered sound even when listening off-axis.

A proprietary lowfrequency driver cone design offers superior damping that minimizes sonic artifacts and unnatural resonance.



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Blue RBN

Peavey unveil ribbon-based powered speakers


ibbon tweeters are widely used in studio monitors, prized as they are for their low mass, fast response and smooth, detailed sound. However, their physical fragility, sensitivity to being overdriven and requirement for an additional transformer stage mean that they are rarely first choice when it comes to live sound reinforcement. Peavey are out to change all that, however, with the RBN 112, a powered PA speaker equipped with a 120mm ribbon driver. Peavey’s proprietary ‘true ribbon’ HF driver is mounted in a specially designed waveguide. Coupled with the integral amp and DSP designed into the speaker, Peavey say they have overcome the usual weaknesses of this type of driver, promising maximum reliability and smooth high-frequency performance without the tendency towards harshness that compression drivers can exhibit. Accompanying the ribbon (which has its own switchable blue backlight, should you wish to show it off) is a 12-inch, dual‑voice‑coil, neodymium Scorpion woofer. The RBN 112 boasts 1500W of power and two separate XLR/TRS inputs, each with a high-pass filter, nine-band graphic EQ and delay. Thru and subwoofer outputs are provided, with DSP compression and crossover control. The RBN 215 subwoofer, which features a pair of 15-inch woofers and 2000W of power, is designed to partner the RBN 112. Both the RBN 112 ($1499.99) and RBN 215 ($1999.99) are available now. Peavey +1 877 732 8391

Can controllers

Sterling launch cost-effective headphone amps


terling Audio have introduced a trio of affordable new headphone amplifiers. Featuring simple controls and gold‑plated input and output jacks, Sterling say these no‑nonsense devices will deliver reliable, low-noise performance. The four-channel Sterling S104HA is the simplest and most compact of the three, with a stereo input at the rear on a single quarter-inch TRS jack and four stereo headphone outputs on the front panel. Four faders on the top panel control levels, and the unit is powered by an external 12V DC mains adaptor. The four-channel S204HA takes a different approach, with four front-facing volume knobs and a tough rubberised casing around its lozenge-shaped aluminium enclosure. The four headphone outputs are at the rear, along with left and right TRS line inputs. There’s an additional unbalanced stereo input on the front panel and the unit is again powered by an external adaptor. Stepping things up considerably, the S418HA is a 1U rack with eight discrete stereo amplifiers and an internal power supply, all squeezed into a steel chassis. Each of the eight headphone outputs on the front panel has a level knob, level meter, source selector and mono/ stereo switch, all backlit for visibility. The two main balanced stereo inputs are at the rear, along with balanced parallel outputs, and there’s a level control for each main input at the front. The eight headphone outputs can also be fed from individual direct inputs, with parallel outputs at the rear to feed additional headphone amps. The S104HA ($49.99) and S204HA ($79.99) are available now, while the S418HA ($149.99) will be available from September. Sterling Audio +1 888 621 2154

Ringing the changes

Waves develop automatic feedback eliminator plug-in


romising automatic and fully effective feedback elimination, X-FDBK is the latest live‑sound plug-in from Waves. Designed as a one-stop solution to the fiddly process of ringing out the PA and stage monitors, this utility looks to not just make the process much faster and easier but also more precise, surgically targeting the problem frequencies so you can maximise gain without delivering an overly mangled and muted sound. To use X-FDBK, you simply turn up your levels until feedback occurs then let the plug-in do its thing, quickly identifying the frequencies where feedback is occurring and applying the necessary notch filters. These settings can then be adjusted, and you can add more filters manually if required. Another advantage is that, rather than requiring two engineers —

one on stage trying to create feedback, the other behind the desk applying EQ — a single engineer on stage with an iPad can do it all. X-FDBK is available now for Mac and PC. It supports MultiRack, StudioRack and eMotion platforms, plus all major DAWs. It costs $149. Waves Inc +1 865 909 9200

July 2016 / w w w . s o u n d o n s o u n d . c o m

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Fender Fortis F‑12BT Active PA Speaker Fender’s smart new PA range offers ease of use, plenty of power — and built‑in Bluetooth music streaming. MIKE CROFTS


he new Fortis range from one of the music industry’s best‑known manufacturers currently consists of two full‑range speakers, based on a two‑way format with the option of 12‑ or 15‑inch woofers, accordingly designated F‑12BT and F‑15BT, respectively. The 12 or 15 part is obvious enough, but the ‘BT’ suffix is there because these speakers are Bluetooth‑equipped, which is both an interesting prospect and an obvious step to take in the world of portable live‑sound gear. I spent a while using a pair of the smaller of the two models, the FB‑12BT, and I was able to give them a decent enough workout in the workshop. I also got to take them out on an event and use them at a couple of band rehearsals.

terms I’d be just as happy to use them for an upmarket AV event as for a pub gig. The cabinet of the F‑12BT is made of nine‑ply 15mm thick “voidless plywood”, according to Fender’s specifications, and is finished in an attractive, non‑reflective textured black coating, which appeared pretty tough and durable

Holding The Fortis First impressions really do count, and the Fortis speakers arrived well packaged (not necessarily something you can take for granted, in my experience) and protected against transit damage in their original Fender‑branded shipping boxes. I liked the look of the Fortis speakers; they are not the smallest 12‑inch boxes out there, but they have a reasonably substantial look and feel about them, and in visual July 2016 / w w w . s o u n d o n s o u n d . c o m


— an impression which was empirically reinforced after I accidentally bashed one against my workshop wall. The front grille is full‑face, powdercoated steel with an opaque black backing of some kind, which looks very good. I find most corporate clients prefer not to see the driver components anyway, and the extra layer of protection against the ingress of dust, dirt and unsolicited liquids is a most welcome feature. Sticking with the cabinet for the moment, the Fortis is well‑endowed in the handle department, with three full‑size cup‑and‑bar handles on both sides and the top. Consequently I found no difficulty in picking it up from any position and carrying it around — an important consideration for potential owner‑operators or those (myself included) with increasingly limited roadie options. The Fortis speakers are designed to be used as floor monitors, if required, and there are substantial rubber bars which act as good solid feet on two sides; as the cabinet shape isn’t symmetrical this means that you can operate the Fortis at two different monitor angles as the occasion demands. Standard M10 flying points are installed, and on the bottom of the enclosure there is a dual‑angle pole‑mount socket. This is a useful feature (and one still not provided as standard on some makes of portable live-sound speaker) that is great in smaller venues, for example, where the high frequencies benefit from being directed 7.5 degrees downward at a nearfield audience, which can be especially effective for speech reinforcement. The 12‑inch version of the Fortis weighs an

Fender Fortis F‑12BT $649 PROS

• Versatile. • Easy to use. • Bluetooth connectivity. CONS


A capable and affordable unit that sounds good, is built to last, and has a useful wireless music-streaming facility built in.

easily manageable 19.5kg (that’s 43 of your classic avoirdupois pounds), and it’s pretty easy to lift on to a stand because the large recesses around the handles allow a good grip without hurting your knuckles. This may seem like a trivial point, but it tends to assume a greater significance after several rounds of setting up (and down) or if you’re intending to play guitar straight after! Inside the cabinet is a 12‑inch woofer, which has a pressed‑steel basket and a large ferrite magnet structure, and a one‑inch‑exit compression driver mounted on a moulded horn assembly with a fairly thick layer of acoustic damping material around the enclosure. Should driver replacement ever become necessary (a most unlikely event, it has to be said, what with all the onboard active electronics managing and protecting your investment), you’ll be pleased to know that the drivers are wired up with good‑quality spring‑loaded connectors (no soldering), and the HF driver at least is a readily available item, according to its label. Call me sad, but this is something

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I like to consider when adding equipment to my own inventory!

Making Connections The F‑12BT control panel houses all the in/out connections and EQ controls (of which more later), and is large, neat and clearly designed. Anyone who has even the faintest idea of what a self‑powered loudspeaker does should be able to connect this up and get it running within a minute or two, and there’s something reassuring about the amount of space between the controls and the almost modular appearance of the input control sections.

“I used them for vocals (male and female) and keyboards, and then tried one as a main speaker and the other as a floor monitor, all with pleasing results.” There are three inputs which can be mixed together, and which offer connectivity and source level options. The one at the top is labelled Line In and has only a ‘combi’ jack/XLR input connector, a rotary level control and a high/low button to accommodate different incoming line signals. In this case the ‘high’ setting reduces input sensitivity, so the choice of ‘high’ or ‘low’ refers to what you’re feeding into it, not the sensitivity of the input stage. The second input is exactly the same, except that on this channel the input button is for line or direct microphone input signals (the

line setting is the same as ‘high’ on the first channel). Now at this point I would normally comment upon — nay, lament — the omission of a pair of unbalanced RCA phono inputs on the line channel, which would enable the solo performer to connect a media player for backing purposes and use a mic straight into the other input... However, the Fortis speakers do have a stereo mini‑jack input on channel three for this purpose, and are also bang up‑to‑date in this department by including Bluetooth capability as well. The issues I have experienced with any form of wireless connectivity between pieces of live sound gear have been numerous, but I was pleasantly surprised when the Fortis’s Bluetooth facility just, well, worked. I pressed the pairing button, which resulted in my iPad Air 2 finding the Fortis very quickly and automatically establishing a connection, and I was able to stream material from all areas within our warehouse — that’s over 15 metres away — with no interruptions at all. Dropouts began to occur only when I went outside the (metal-clad) building, shut the door, and stood on the other side of my van — and as soon as I was within range again the Fortis connection was re‑established and streaming continued without me having to reinitialise anything or restart the iPad playback. The auxiliary input on channel three, be it via the mini‑jack or Bluetooth, can be summed to mono or maintained as a stereo feed, in which case the connected speaker will amplify the left channel and route the right channel to another speaker via an ‘extension’ XLR output. It will also provide a local mix output in mono from the ‘loop’ connector, so just about every option is

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catered for. It’s a well thought‑out system and should prove both popular and useful (but I’d still have prefered a pair of RCA inputs over the mini‑jack).

The Roaring Fortis Among other things, I like to wind up new speakers to see what happens when they are running flat-out, and I do this in three different acoustic spaces in addition to performing a road test. For obvious reasons it isn’t always possible, when doing a village hall event, to bang out pink noise at full power for very long

before the caretaker or neighbours start to fret, so the studio is always a good place to start, followed by the somewhat more reverberant space in the warehouse loading bay and finally in the open fields out the back. The Fortis 12‑inch and 15‑inch versions both have the same quoted maximum output level of 135dB SPL, so I did a quick (if unscientific) measurement at one metre on the HF axis in the studio, with a few extra absorbent panels clustered around, and then again in the open air. At this point I discovered that my ‘big’ sound level meter had succumbed to horrifying corrosion within the battery compartment and my backup unit only measures to 130dB... However, I can report that on the A‑weighted setting (with pink noise both on crescendo and burst, indoors and out), I measured 130.9dB along with the word ‘over’, which supports my view that these modest‑sized portable speakers do in fact play pretty loud. As for the output quality, that’s always subjective and depends to some extent what you want to use them for, but I did a lot of playing around with the EQ settings and found that there’s enough variation across the different settings to achieve a noticeable degree of variation even at high output levels. The mid/ high EQ has a ‘high’ setting, which really does boost the top end and might be useful for speech applications, although I couldn’t imagine needing that much brightness unless, say, I’d had to put a bin liner over the speaker on a rainy day. The ‘mid’ setting on the other hand acts on predominantly vocal frequencies, but also adds a distinct emphasis to the lower mids that, on appropriate material, adds

July 2016 / w w w . s o u n d o n s o u n d . c o m


a lot of punch and could easily be labelled ‘funk’... The LF EQ settings offer a boost curve (the Fortis has a good full bottom end anyway, so I’d imagine this wouldn’t be needed often), and two LF reduction settings — one LF shelving curve for floor monitor use and the other called ‘sub’, which is a high‑pass filter for when the Fortis is used with an external subwoofer. I found that the FB‑12BT had a big, warm sound which belied its 12‑inch format and overall cabinet size, although I felt that using the mid‑boost EQ kept a better balance with various styles of music signal, particularly of the ‘punchy pop song’ type. For playing in larger spaces, the full potential of the FB‑12BT can be unleashed when coupled to a suitable subwoofer, and the Fortis’s considerable output power capability can be focused on the material above say, 90‑100 Hz. By way of a road test, I used the speakers at a local dinner/dance event with recorded music and lots (and lots) of speeches, and everything was loud and clear above a boisterous audience of around 200 in a ‘typical’ village hall. At a band rehearsal I used them for vocals (male and female) and keyboards, and then tried one as a main speaker and the other as a floor monitor, all with pleasing results. They were easily loud enough to get over the band, and with a nice full vocal balance. Needless to say I kept well away from the HF EQ setting, as per my previous comments After testing and using the Fortis speakers I am left with the impression that they are capable performers that are very easy to transport, rig and use effectively in a range of applications. The Bluetooth facility came in unexpectedly handy at

Alternatives There aren’t many active PA speakers with Bluetooth streaming on the market at the moment, but the Turbosound iP and iX series have it, as does the Alto TS212W.

the dinner/dance, as I was using a remote digital mixer (the sort where everything is contained within a digital stage box), and the one drawback of this system is that you have to connect playback tracks at the stage end rather than where you’re mixing from; as the distance was well within range I used my iPad again, and was able to cue and control the music from front of house using the Bluetooth connection. Neat!

Life Begins At Fortis The Fender Fortis F‑12BT is a versatile powered speaker with added flexibility offered by its Bluetooth connectivity. It looks the part, has more than enough output power for small and medium‑sized venues, and can maintain a smooth and well‑balanced sound when pushed fairly hard. It is very easy to set up and operate, and is likely to be an interesting prospect for travelling bands, music venues and solo performers. The 12‑inch version I tried sounded full and warm, and there was plenty of bass available for playback of recorded music. I like the ‘timeless’ styling, which would fit in well in almost any venue, and I can easily imagine these speakers on the shortlist for installations in pubs, schools, community halls and houses of worship. $$ $649.99 per speaker. TT Fender +1 480 596 9690 EE WW

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Alto Professional

MixPack 10

Powered Mixer & PA Speakers For some gigs, less is more — and that’s where this simple, affordable and portable all‑in‑one PA system comes in!



f you get involved in any way and at any level with portable live sound, there’s almost certainly going to come a time when you just want something small, light and, above all,

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simple to use. The MixPack 10 from Alto Professional aims to provide enough functionality to be capable of handling a modest‑sized group performance, but in a small, light and easy‑to‑use format which will appeal to anyone who is in need of a ‘small PA’. The MixPack 10 is, in essence, a highly portable complete PA system comprising an eight‑channel powered stereo mixer (ie. with built‑in power amps) and a pair of passive two‑way speakers. While the MixPack 10 design focuses on portability, there is enough input-mixing capability and output power to handle more than your usual ‘solo busker’ type all‑in‑one boxes. I’m sure that this simple little system will be of interest to many potential users who are not regular sound guys (I get more hire enquiries for this type of gear than anything else), so I’ll take a fairly detailed look at the MixPack 10 in terms of its functionality and ease of use.

Unpacking The MixPack 10 consists of three units — two speakers and a mixer — and is supplied with all the necessary power and speaker cables. The two speakers are, as far as I can tell, identical: they are made of injection‑moulded plastic and have a dark grey matte finish, with a metal grille covering the woofer and an integral waveguide moulded into the cabinet for the HF exit. The driver format is standard two‑way, with a 10‑inch LF unit and a one‑inch‑exit compression driver (a ‘proper’ one, not a piezo device, which, at this size and price, is a tick in the box for Alto). The crossover point between the drivers is

Alto Professional MixPack 10 $399 PROS

• Small, light and easy to carry. • Very easy and quick to set up. • Simple control surface but with up to eight inputs. CONS

• Not the most powerful system for its size, though it does the job. SUMMARY

A neat, simple and cheap sound system that even the most inexperienced musician/engineer can use straight away. Live sound doesn’t get much simpler than this!

2.2kHz and, although the HF dispersion (coverage angle) isn’t mentioned in the specification, the coverage seems to be fairly wide so I’m guessing it’s designed to be around a nominal 90 degrees in the horizontal plane. The speakers are neat and compact, and although they don’t have (or need, for that matter) handles on the sides they do have a ‘top slot’ handle which serves well enough for carrying them around. The speakers are so light, at only 6.8kg apiece, that mounting them on a tripod stand or pole is easy, even at operating height. The moulded enclosures seem sturdy and well‑finished, and are equipped with standard 35mm pole‑mount sockets on the bottom; they also have what look at first to be potential suspension points but on closer inspection (I removed one of the blanking plugs) are simply holes in the cabinet, presumably intended to be used

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for threaded inserts on other models which use the same enclosure moulding. The cabinets are shaped with an angled surface so that they can be used as floor monitors, and there’s an obvious setup option for a solo performer to use one speaker channel for the front of house and the other as a wedge monitor — although there are limitations to this, as I’ll described shortly.

Mix It Up The heart of the MixPack 10 system is the mixer itself, which is a very compact unit that is designed to sit inside one of the speakers. To mount the mixer all you do is locate a couple of metal locating tabs at the bottom and secure the top with a large and easily operated rotary latch — it takes about a second, and removal requires only a push‑and‑turn action which is also easy and quick. The system can be used with the mixer in place or removed and operated remotely, the only connection to the speakers being a couple of jack‑to‑jack cables which are supplied in the retail box; there’s also a very short speaker cable supplied for when the mixer is nested inside the cabinet. The MixPack 10’s mixer is very small and light, and is built into a strong, neat and well put‑together steel housing. I assume that the MixPack 10 is so‑called because of the speaker format, as the mixer has eight input channels: four mono mic/ line inputs and two line‑level stereo inputs. It’s a simple mixer, but it has all the necessary vital ingredients. The four mono input channels have both XLR (mic) and TRS (line) connectors, and the XLR inputs have a globally switched

48 Volt phantom-power option. Each mono channel has a rotary level control feeding the mix bus, and a two‑band EQ circuit that applies up to 15dB of cut or boost centred on fixed frequencies at 12kHz and 80Hz. The EQ seems to suit the system, especially the LF frequency control, which is set at a low enough frequency to avoid too much booming and boxing in the mid-range if it’s turned up (as I suspect it may well end up being) for a fatter sound. The lack of any real mid‑range control is something I’d normally worry about, but when you put it in the context of the overall system and what it’s designed to do, I didn’t find it a problem.

“The MixPack 10 is one of the most convenient and easy‑to‑use portable PA systems I’ve come across...” The stereo channels are equipped with TRS sockets (channels 5 and 6) and unbalanced RCA‑type phono jacks (channels 7 and 8), and have the same two‑band EQ. The input jacks on channels 5/6 can operate in either mono or stereo: if an input is only connected to the left input (5) it will appear on both left and right outputs. None of the mixer channels are provided with any kind of input trim, so you have to be careful not to drive them with too hot an incoming signal. There isn’t a clip warning LED, either, so input level is something that inexperienced users will have to keep an eye (and ear) on.

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So, basic mixing is just about as simple as it gets, with no input trim to set up(!) and good old bass and treble controls for the EQ. The output level is indicated by a small LED ladder meter, and so long as the output sounds all right and the red limit segment isn’t glowing, everything should be going well enough. The MixPack 10 provides a ‘monitor mix’ that is a mono sum of the main left and right output buses, at line level, for connecting to a self‑powered monitor speaker or another system. The monitor level can be controlled independently of the main output but the mix is always the same; a nice touch here, though, is that the monitor signal appears on two jack outputs. This means that a couple of monitors can be connected without needing to daisy-chain them together, or you can feed two different systems. In terms of the main mix, none of the mono channels can be panned left or

right, so the only stereo in the output will be anything that’s connected to the two stereo input pairs on channels 5 to 8. This isn’t an issue in normal use, but it does impose some limitations if, for example, you wanted to run one speaker output channel as ‘front of house’ and use the other speaker as a monitor: you would (a) get exactly the same mix and level in each speaker, and (b) if a true stereo source were connected you’d get one channel in the main and the other in your monitor. I should point out that the manufacturers don’t suggest using the MixPack 10 in this way, but it seems like an obvious possibility for solo performers in small spaces. The MixPack 10 has a built‑in effects processor, which can be applied to any or all of the four mono input channels; it is simply switched on or off for each (there’s no adjustment for the send level) and is controlled by an overall master

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level pot. The effect itself is described as ‘reverb’, although to me it sounds more like a delay/echo, and is there should you feel the need... I can think of one regular customer who would dial this in on his mic every time! A button next to the main output level control offers a global EQ setting for speech use, and under some circumstances this can be very useful for removing unwanted frequencies and in effect emphasising the vocal range. This very compact unit houses mixer and power amplifier, and I like the presence of a built‑in cooling fan; on occasions where the mixer is used inside the speaker there’s not really any cooling air able to circulate around its casing, so this is a plus point for reliability.

Fire It Up Setting up the MixPack 10 is a breeze, and from opening the car boot to saying ‘one, two’ should take less than five minutes. Provided you don’t try to drive it too hard or expect more output level than it’s comfortably capable of in any given space, it sounds perfectly decent, and there aren’t enough knobs and buttons on it for anyone to get anything horrendously wrong. Set the EQ controls to flat, turn the reverb right down (to begin with), set the main level in the middle, then turn the inputs up one at a time until they are balanced and sound clean, and you’re done. This isn’t designed to rock the foundations of a large venue (the specified maximum output of 113dB SPL at 1m gives some idea) but it’s loud enough for a party PA and more than enough for bars and small venues. I can tell you it works

Alternatives Similar ‘all‑in‑one’ solutions include the likes of Yamaha’s MSR 250, the Behringer EPS500 and the Superlux SP108.

great outdoors for announcements and background music at your local Summer Fair, because I used it twice in this setting. Just out of interest I replaced the original Alto speakers with a couple of other types from my stock, and all things considered, I’d stick with the supplied units as the whole thing is designed to work well as a complete system. I do like the monitor output facility, though, because this is an easy way of upscaling for the odd larger event, still using the same familiar mixer but linking in to a more powerful powered system.

Conclusions The MixPack 10 is one of the most convenient and easy‑to‑use portable PA systems I’ve some across, and there’s nothing — apart from possibly overdriving the inputs — to confound even the most inexperienced or infrequent user. The physical format is small, light and very easy to transport, and even the cables can be stowed inside whichever speaker isn’t currently accommodating the mixer unit. With a street price just north of £300 the MixPack 10 has got to be worth a look if bells and whistles aren’t really your thing, and you favour an easy life. $$ $399. TT InMusic Brands +1 401 658 3131 WW WW

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iP1000 Column PA System

Turbosound are no strangers to the world of big-box PA systems, but how good is their new compact column speaker? PAUL WHITE


onceptually, Turbosound’s new iP1000 PA system follows the increasingly familiar format of a tall, thin ‘line‑array’ style speaker column sitting atop a subwoofer. Though this type of speaker is commonly referred to as a ‘line array’, technically it is a hybrid design, as there’s only one tweeter. However, the waveguide is designed to match the dispersion of the main drivers, and this is a very similar approach to the one used in my own Fohhn system. Turbosound have, of course, been a part of Music Group — the company that owns Behringer, Klark Teknik, Midas, Tannoy, Lab Gruppen, TC Eelectronic, TC-Helicon, Bugera and a growing list of others — since 2012, and what that means in practice is that various technologies can be shared between the brands. More specifically, the iP1000 incorporates Klark Teknik’s mature and highly regarded DSP and amplifier technology. Manufacturing of the iP1000 is handled at Music Group’s own factory complex in China, which is how the product can be offered at such an attractive price. The Fohhn system I use myself is of a very similar size, but the cost of the iP1000 would barely pay the VAT on mine! Aside from keeping visual intrusion to a minimum, such systems tend to be portable (this one weighs just 24.8kg), they have a very wide horizonal dispersion (up to 120 degrees, in this case), and because of their use of multiple small drivers, the crucial mid-range speech register is covered with more clarity than you might expect from a typical ’12-inch woofer July 2016 / w w w . s o u n d o n s o u n d . c o m


plus tweeter’ arrangement. In the iP1000, the mids and highs are covered by eight neodymium drivers specifically designed for this system, each just 2.74 inches in diameter, teamed with a one‑inch, horn‑loaded ‘super tweeter’. All the drivers are housed in the upper 60 percent of the single-section aluminium column, and they are protected by a powder-coated, perforated-steel mesh grille. The column locates into the top of the sub enclosure via a multi-pin connector and four locating bars, so all you have to do is drop the top in place and you’re good to go. All the electronics are packed into the dual eight-inch subwoofer, and include 1000 Watts of Class‑D amplification fed from a switched-mode power supply. A DSP section provides a three-channel digital mixer, presets for system EQ, the active crossovers and also safety protection, including separate HF and LF limiting. Birch ply is used for the black‑finished subwoofer housing, and there’s a grab handle built into the top. Mains comes in on the usual IEC socket with adjacent power switch, though unlike many switched-mode systems, this one is not universal voltage. By way of performance, the iP1000 has a frequency range of 50Hz to 20kHz (±3dB), and can generate SPLs of up to 122dB. Its 120‑degree coverage angle is measured at the -6dB points.

Going In Audio may be streamed to the iP1000 via BlueTooth, and there’s a free app that can be used to control the mixer (though it can also be controlled directly from a panel on the sub, as there’s little to adjust other than the channel levels and overall EQ). When using a single iP1000, the streamed audio

Turbosound iP1000 $599 PROS

• Compact and manageable. • Clean sound for both vocals and music playback. • Adequate level for a system of its size. • Free control app. CONS

• There’s no separate channel EQ on the built-in mixer though I’m guessing most users will plug in their own mixer with effects. SUMMARY

A solid performer at a surprisingly attractive price.

is summed to mono, but when another ‘slave’ iP1000 is connected to the ‘master’ speaker’s Link XLR output, the BlueTooth can be configured to operate in stereo. The integral mixer offers two ‘combi’ jack/ XLR inputs that can accept balanced or unbalanced signals (there’s no phantom power option) at mic or line/instrument level, and the control panel uses a rotary encoder with integral push switch to select and then adjust the various functions as shown in the backlit, blue 128x32 LCD window. There are four switches marked Process, Setup, Exit and Enter, and the operation is so intuitive there’s no need to look in the manual. There are four ‘System EQ’ voicing options, called Music (actually a flat response, and the setting I used for most of the tests), Live (bass boost), Speech (mid boost) and Club (a ‘smile’ curve), and in addition to these there’s a separate Position EQ setting to optimise the low end for

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when the speaker is placed near walls or corners. There’s a further global three-band EQ, offering simple bass, middle and treble control, and input gain control is provided for the analogue and Bluetooth inputs.

Setting Up When assembled, the system stands just under two metres tall, with the sub being only a shade over 260mm in width. The iOS app is designed to automatically detect and self-configure for mono or stereo master/ slave setups once the BlueTooth pairing has been taken care of. However, having only a single system for test, I can only vouch for its mono operation. The app’s GUI is very

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simple, with just one page showing the level and EQ controls and a second the global system parameters. The manual for the system is very basic but it does get you up and running in minutes. Setup is simple but, like all the similar systems I’ve looked at, the column is only supported at the base and probably wouldn’t take kindly to a drunken reveller

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Alternatives The nearest alternative is probably one of the HK Elements systems though LD Systems also make an affordable compact line array. JBL also announced one earlier this year, but we haven’t yet had one to test.

Performance barging into it. The manual says that the dedicated control app is free from the App Store, but it stops short of actually telling you what it’s called. All my attempts to search for it in the App Store brought up driving games with ‘Turbo’ in the title but no sign of the app I wanted, so I accessed it from the Turbosound web site which obligingly locates the Turbo Control App for you in iTunes. The BlueTooth paired very readily and the free app accesses all the control functions you can get to from the sub’s control panel, but doesn’t appear to offer anything extra.

With the system set for a flat response, the tonality when using a good dynamic vocal mic was clean and clear, with the Voice preset adding a mid hump that could be useful for some events. The Live and Club settings also work as expected, and these can be tempered using the fixed frequency three-band EQ if desired. Music playback usually shows up how poor some budget PA systems actually are, but the iP1000 did a very creditable job, leaning much more towards a ‘hi-fi’ sound than most similarly-priced conventional boxes. There was also plenty of level, with a nice solid bass but without excessive boom. The sub level can be adjusted independently of the top, but the default setting seemed about right to me. A pair of these should be quite loud enough for a pub band PA, while a single unit would work well for solo performers, duos or even for wide dispersion monitoring. In common with other compact linearray-type systems I’ve tried, the horizontal dispersion is usefully wide while the narrowed vertical dispersion helps keep from spraying sound onto the floor and ceiling. Given the performance and portability of the system, not to mention the very attractive price, I have to say that I’m impressed. And if you need a hint more power and bass extension, there’s always the slightly larger iP2000. $$ Mono system $599. WW

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The Southern Wild: our engineer reminds us that you don’t have to cram every mic signal into your mix just because it’s available. SAM INGLIS


here’s something special about recording a group of musicians all together in one room. You might not capture the most technically perfect performance from everyone, but any deficiency in the parts is made up for by the excitement and naturalness of the whole. It’s by far the quickest and most engaging way to capture what a band is all about. At the same time, though, recording ‘live in the room’ introduces technical challenges, especially when the room in question belongs to a house rather than an acoustically treated studio. These are compounded when the band

are self-recording: what might be obvious to an engineer listening in an isolated control room is easy to overlook when you’re focusing on your own playing and singing. So it was that Americana three-piece the Southern Wild had emerged from an enjoyable recording (and filming) session with a multitrack that was proving frustratingly hard to mix, even though all the individual elements were cleanly recorded.

Mics Galore When you’re not in a position to monitor objectively, or to spend hours working on mic placement, there’s a temptation to put up every mic you can lay your hands

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on in the hope of scoring at least some hits. Listening to the multitrack for ‘Curtis Stirling’ suggested that something of the sort had gone on in the Southern Wild’s sessions. Although the song featured only four instruments and one vocal, there was a total of 12 audio tracks, with both bass and guitar represented by DI and two miked tracks, and cajon miked both inside and out. Having recorded lots of mics, there’s a natural tendency to want to use them all, and I wonder whether that might have contributed to some of the difficulties singer Drew Stephenson was having with his own mix. He’d got a pretty good basic balance of instruments, but there was a general muddiness to the sound and more than a whiff of comb-filtered, roomy spill, suggesting that too many ingredients had been thrown into the pot.

Don’t get me wrong: there are times when double-miking a source can be very worthwhile. Most obviously, of course, it’s a basic requirement if you want that source to have any stereo width. Some engineers also do it in order to exploit the tonal properties of different microphones, or to capture important but separate elements of the sound produced by large instruments such as the double bass. However, a lot of people put up multiple mics with the rather more vague goal of “having options available at the mix”. This is a mixed blessing to say the least, and all too often it produces two or three unsatisfactory sounds rather than one really good sound. I’m not sure how the Southern Wild arrived at their mic choices and placement but, in the end, I don’t think they gained much by leaving these

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options open. Neither of the two mics on the bass amp had managed to obscure its fundamentally honky and cheap sound; and the two mics on the cajon had been placed so close that neither really sounded much like a cajon. Drew’s acoustic guitar had been miked in stereo, which was a reasonable idea, but both mics had also picked up a great deal of bleed from his vocal and the other instruments.

The One & The Many When you’re faced with a combination of room and close mics, I find it’s quite often the case that each individual source needs to favour one or the other. That is, for any given instrument, the close mic can supply the main sound with support from room mics and spill, or the close mic can be used to subtly assist that instrument’s presentation in the room mics. Problems tend to arise more frequently if an instrument’s presentation in the mix comes equally from two or more different mics. With this in mind, I began the mix by trying to identify one track in each case that would provide the main source for

“In my experience, mix-bus processing is more useful for shaping the tone of the mix than for dynamic control, the general aim being to fill out every last corner of the mid range.” each instrument. Listening first to the room mics (see box), it was clear that the band’s makeshift acoustic treatment had made their recording space sound pretty decent, at least down to 300Hz or so. However, the room mics hadn’t picked up the sort of balance of instruments you’d want in the final mix: the cajon was much more prominent than the other sources. It also sounded much better in the room mics than on either of its close mics — some instruments just need a bit of space for their sound to bloom and come together. The balance in the room mics wasn’t quite so skewed to the cajon that I could get away without using its close mics at all, but this was definitely a case where they would be best off operating in

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a supporting role. However, figuring out the best way to make this work wasn’t as straightforward as I’d hoped. Somehow, the close miking had produced two tracks which sounded more like abstract electronic beats than a plywood box being struck with fingers and palms. Both mics also emphasised a very strong resonance around 120Hz. I managed to tame the resonance by time-delaying the cajon’s inside mic by about 3.5ms, so that a helpful phase cancellation occurred when it was combined with the outside mic. This improved the sound of the cajon mics in isolation, but I wasn’t going to be using them in isolation. The cajon sound in the room mics had plenty of snap and buzz; what it lacked was mid-range punch, so I added some fairly heavy processing to the close mic bus to focus the sound on this missing region. One compressor and two distortion plug-ins later, I had an overall cajon sound that both sounded like a cajon, and had some substance and power to it.

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Up Close The cajon’s prominence in the room mics made clear that for all the other instruments, it would need to be close mics or DIs that took the lead role. But which ones? It didn’t take much listening to convince me that that bass sound would have to come entirely from the DI, as the amped sound was plain bad. In all honesty, the DI’d sound wasn’t amazing either, but it was at least clean, spill-free and generally amenable to being improved. It took quite a bit of compression to bring the dynamics into line, and I found that instead of a conventional amp simulator or distortion effect, FabFilter’s multiband Saturn saturation plug-in did a better job of flattening out the sound and removing the remaining honkiness. With a heavier heart, I also decided that the DI would have to carry the bulk of

the acoustic guitar sound, as there was simply too much vocal spill on the guitar mics. Pickups in acoustic guitars can be pretty hard work at the mixing stage, but this one sounded reasonably natural, at least once I’d scythed away the usual excess of hard-sounding mid range. Being a sentimental sort, I kept in one of the guitar mics at a relatively low level; I’m not sure its contribution was really noticeable

Careful With The Sides One of the challenges Drew Stephenson had faced in setting up the Southern Wild’s recording session was that he didn’t have a good matched pair of microphones. Very sensibly, therefore, he had chosen to set up his room mics using the one stereo technique that doesn’t require matched mics, namely Mid-Sides. Acknowledging the importance of the room mics in a live recording, he’d reserved his two ‘best’ mics for this role: an SE Electronics SE2200 capacitor mic for the Mid and a custom Xaudia ribbon for the figure-of-eight Sides. This was a good decision, but it meant that there was quite a contrast between

the tonality of the two, with the bass and low mids much more strongly represented on the Sides track than on the Mid track. And this, in turn, meant there was a lot of muddy, out-of-phase low end on the decoded left/right stereo room track, which is something to be wary of at mixdown. It’s very hard to predict how such a ‘wide’ bottom end will behave in any given listening environment, and it will simply disappear when the track is auditioned in mono, so in my remix, I high-pass filtered the Sides track fairly savagely. To try to match the tonality of the two mics a bit more closely, I also boosted the Mid’s mid and the Sides’ high frequencies.

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Remix Reaction Drew Stephenson writes: “As this was the first time I’d tried to record and mix from multiple instruments I wasn’t really sure how to get started. I knew I wanted to try and capture the feel of the three of us playing together in the room, so I tried to focus on using the room mics as much as possible and then bring up the close mics as needed. One of the challenges was that, having set up multiple mics to give myself plenty of tone options, I then tried to use all of them... “Getting the tone balance right in the end, but I convinced myself that it made the sound a little more natural. There was no choice on offer when it came to Drew’s vocal, but there was a need to do something about the way it sounded. He’d sung into a Shure Beta 87 stage mic which, I suspect, had been placed a bit too far away, because the basic tone was thin and harsh. This is a problem I’ve had to address in several previous Mix Rescues, and I’ve found it often doesn’t respond well to static EQ, because not every word or phrase is equally affected. A multiband

between the bass and cajon was particularly difficult as these aren’t instruments I’m as familiar with. “Obviously we’re much happier with the remix; the sound is fuller without being congested, the bass and cajon occupy their natural frequency areas much better and the dynamic range is much better. I also like the fact that the vocal sounds a lot more like me! And I now have a good template to work with for the other songs from the session.” compressor is usually more effective, and in this case, it took five bands to bring Drew’s vocal into line. All but one were set flat, but with hair-trigger threshold values so they’d duck various bits of the mid range at the first hint of harshness. The lowest band was also configured as a compressor, but when no compression was taking place, it applied a generous boost centred around 400Hz to thicken up the basic sound. On top of that, I found I needed to automate a 250Hz EQ band to top the low mids up in some sections; and though I’d

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Power Trio Panning One of the classic mixing conundrums is that some band line-ups just don’t lend themselves to stereo panning. The convention that drums, bass and lead vocals should be panned centrally is so ingrained that any departure from it sounds like a perverse attempt to return to the late ’60s. But if you’ve only got one other main instrument, that leaves you with a dilemma. Do you pan the extra instrument to one side, keep the core tracks in the middle and accept that your mix will be unbalanced? Do you put everything straight down the middle and mix more or less in mono? Or do you try to compromise the Holy Trinity of centrally panned sources in order to have things out wide on both sides? This was a dilemma that definitely presented itself in the Southern Wild’s recording. Drew Stephenson’s vocal was the most important thing in the mix, and obviously needed to go straight down the middle. But which of the instruments could be panned away from the centre? used a conventional compressor as well, it was no surprise to find that both the vocal and instruments needed some fader automation to maintain the right balance through the song. The remaining instrument was a shaker overdub, which Drew had found hard to place in his mix. Over the years I have come to be increasingly brutal in processing this sort of percussion, which has a tendency to sound spiky and thin. In this case, I led the assault with a huge EQ boost in the mid range and a fairly heavy degree of saturation. It soon waved the white flag and sat meekly in the track.

The bass? The cahon, which was the main rhythmic element? Or the guitar, which, conceptually at least, belonged in the same position as Drew’s voice, since he was playing it? The purist approach would be to maintain the panorama that the room mics had captured. In this case, though, they weren’t much help, as they hadn’t really captured one. Even when I temporarily ramped up the Sides signal to exaggerate the stereo width, it didn’t feel that any of the instruments naturally belonged on one side or the other. In the end, I decided that the most natural approach was to hard-pan the guitar one side and the cahon’s close mics the other way. This didn’t in any way match what was going on in the room mics, but in timbral terms, it meant that there was a left-right balance between the two instruments that were carrying the song’s mid range; and unlike a full drum kit, the overall ‘smaller’ sound of the cahon didn’t seem odd when pushed to one side.

Master Full I’ve never managed to train myself to ‘mix into the compressor’ as some engineers like to, but at some point in the process, I’ll usually put one across the stereo bus as an experiment. Sometimes it stays, sometimes it goes, rarely does it apply more than a decibel or two of gain reduction. On ‘Curtis Stirling’, it stayed. In my experience, mix-bus processing is more useful for shaping the tone of the mix than for dynamic control, the general aim being to fill out every last corner of the mid range. Here, I used SoundToys’

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Radiator and Slate’s VTM to add some general warmth, and pushed the 500Hz-3kHz range into a couple of bands of multiband compression to squeeze as much as possible out of it before the mix became tinny or honky. Thankfully, the Southern Wild’s feedback on my first attempt was not of the “Can you cut half a dB at 267Hz on the second triangle overdub?” variety. Drew’s main request was for less reverb on his vocal, so

as not to undermine the feeling that the listener is hearing a band playing in a living room. When the focus in Mix Rescue so often is to overcome the compromises of home recording, it’s nice to embrace the reality for a change! The decision to record live was exactly the right one for them, and even if it introduced a few challenges at the mix, ‘Curtis Stirling’ has a directness and immediacy that no overdubbed studio multitrack ever could.

Audio Examples Original mix Remix The Southern Wild had recorded ‘Curtis Stirling’ live in a domestic room, and the aim was to improve the general mix without losing that live, raw quality.

As you can hear in the first of these examples, of the three different presentations on offer for the bass guitar, the DI was the most promising. The second example is the DI track as heard in the mix. Cajon raw in out both Cajon processed in out both

Room mics raw Room mics processed The room mics had been recorded as a Mid-Sides pair with non-matching mics. As you can hear, the Cajon was prominent in the room sound.

Two mics had been placed on the cajon. In these examples you can hear them each in turn, then together. Note how in the unprocessed example, the combination of the two mics reinforces the resonance at 120Hz. Vocal raw

Bass raw DI mic1 mic2 Bass processed DI

Vocal processed Drew Stephenson’s raw vocal sounds harsh and thin; in the remix, I reshaped its tone using a multiband compressor.

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Part six‑string, part speaker cabinet, resonator guitars are like no other instruments. Here’s how to capture their distinctive sound. DAVID GREEVES


he phrase may now be more familiar to us in a digital audio context, but back in the 1920s, musicians were engaged in a loudness war of their own, and with electronic amplification still a few years away, it was acoustic volume that counted. Though it would go on to dominate popular music in the second half of the century, the humble guitar lacked the volume to really cut it as a lead instrument, and hard‑working strummers were struggling to hold their own alongside horns, pianos and drums. While some manufacturers experimented with larger body sizes, when an enterprising instrument repairer named John Dopyera hit on the ingenious solution of placing a spun aluminium cone inside the guitar to pick up and amplify July 2016 / w w w . s o u n d o n s o u n d . c o m


the vibration of the strings, the resonator guitar was born. Dopyera had not just given the six‑string more volume; he’d created a new class of instrument, with a tone and character entirely its own. Though its fortunes have waxed and waned over the decades, the resonator (or, more properly, the resophonic guitar) has been a continual presence in blues, folk, roots, country and bluegrass music ever since, and the last 10 years have seen a dramatic revival in interest. To help us figure out the best way of capturing the resonator on record, I’ve enlisted the help of two experienced session player‑producers, one on either side of the Atlantic. Nashville veteran Jimmy Heffernan is a well‑respected country and bluegrass player, producer, and teacher who has toured or recorded with numerous artists including Joe Diffie, Brad Paisley and Charlie Louvin. Meanwhile, Michael Messer is the UK’s foremost authority on all things slide, having spent the last 35 years exploring the limits of the instrument, from blues, folk and Hawaiian styles to fusion and beyond.

Shining Examples The reason I’ve tracked down both of them is to cover both sides of a clear divide, between square‑neck and round‑neck instruments. Square‑neck resonators (as typified by the output of the Dobro guitar company and hence colloquially referred to as ‘dobros’) are designed to be played with the guitar positioned horizontally like a lap‑steel or Hawaiian guitar. Widely used in country and bluegrass, these slide‑only instruments are generally played using a metal bar, and their thick square necks

and high string action (the strings sit up to an inch off the fretboard) offer superior sustain. On the other side of the divide, round‑neck or ‘Spanish’ resonator guitars feature a conventional neck and string action and can be held and played just like a regular acoustic guitar. However, these instruments (sometimes known as ‘nationals’, after the National guitar

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company) are closely associated with blues slide guitar, played using a slide or ‘bottleneck’ made of glass, metal or ceramic material. While the dimensions of the guitar’s body, the material it’s made out of and the style of cone it uses all have a distinct effect on the sound produced, much of the difference comes down to these playing techniques — and to the sound the player is aiming for. While modern country and bluegrass dobro players tend to favour a clean, smooth sound, bottleneck blues played on a national is generally a more rough‑and‑tumble affair. Before delving deeper into these different approaches, let’s take a closer look at the instrument itself, to understand where this unique sound in coming from.

Not Just A Guitar Although for all their oddities resonator guitars are reassuringly guitar‑like, Michael Messer is quick to dispel any illusions of familiarity we might have: “A resonator guitar, though it is a guitar‑shaped object and it’s played like a guitar, in many respects it’s not a guitar

as far as sound goes,” Messer says. “It’s more like a speaker cabinet, with the resonator cone acting like a loudspeaker — that’s really what it is. And the way that you would record an amplifier in a room is different from the way you would record an acoustic guitar in a room.” A conventional acoustic guitar sends out sound waves from a variety of places — principally the top, or soundboard, but also the sound hole, the back and sides, and the strings themselves. The resonator guitar adds a further element, the resonator cone, which picks up string vibrations via the bridge and amplifies them, sending the sound up and out of the instrument but also down inside to then emerge from the soundholes. “The body has less influence on the overall sound than on, say, an acoustic guitar,” says Jimmy Heffernan. “The main sound is the sound of the resonator cone, coupled with the ported sound of the screen holes — they’re basically bass ports.” So, while the cone delivers more of the brash, high‑mid content of the instrument’s sound, the ports in the body provide more of the warmth lower down

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mic but, depending on the situation, you can also get wonderful results stereo miking,” says Jimmy Heffernan. “Of course, it depends on context. A lot of engineers, being new to recording dobros, or even new to hearing them, tend to think, ‘Well, it’s a slide guitar and it needs to sound thin’! It’s

the frequency range. Michael Messer agrees with this diagnosis. “There are various types of soundholes on resonator guitars,” he says. “You’ll see f‑holes, grilles, little things that look like tea strainers on dobros. You put the mic up to one of those holes and you’ll get a good bottom‑end sound. Depending on where you put mic (over the cone, over the back of the cone, close, far away, over the f‑holes and so on) you get a complete range of sounds, something it’s actually harder to get from a regular acoustic guitar.”

Favourite Position Where you choose to position your mics is likely to depend on what balance of these various elements you require. For modern dobro playing, the aim is usually a balanced sound reflecting the full frequency range of the instrument. “You can get great results with a single

quite the opposite, actually — it needs to be rich, full‑bodied and not thinned out. Stereo miking does wonders for that. A big feature of country and bluegrass playing is the use of syncopated riffs that alternate between the high and low strings, so you really need to capture the full range.” If you have the luxury of a decently sized, good‑sounding recording space, you’ll be able to capture a more complete picture of the instrument’s disparate sound sources by miking further away. That could mean a single large‑diaphragm

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mic or a stereo pair, but Heffernan also says, “I’ve got good results using a small‑diaphragm condenser up close, aiming at a point between the cone and bass port, and a more ambient large condenser two feet away, capturing a more rounded picture of what I’m hearing, together with the more punchy sound from the close mic.” Michael Messer prefers a single‑miked approach. “I’m not a great fan of using more than one microphone,” he says. “Occasionally I do. There are always exceptions to rules because there are no rules in the studio as far as I can see. It’s trial and error. I’ve been making records for 30 years, but every time I start it’s as if I’ve never made one before. But I do prefer one microphone. I prefer it in the mix, with the sound coming from one point rather than more than one point.”

There is, however, one thing about which both players are in harmony. “When recording resonator instruments,” says Messer, “I would say generally don’t mic too close, as you tend to only get the sound of the picks. You’ve either got plastic or metal picks in most cases, and that can be a nightmare.” “The sound of the picking is kind of built into the sound of the instrument,” says Heffernan, “but you try not to have the mic picking up the picks that much. I would never have a mic anywhere close to the picks. It would be on one side or the other of my right hand. There are very few professional dobro players in

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the world and, almost to a man, they live over here [in the US]. So if you’re recording a dobro player, probably what you’re going to encounter is somebody that has little recording experience. That includes everything from setting up their instrument to the strings to knowing where to play and all that. So your first encounter with a dobro may not be with a professional musician.

“On any instrument, it’s a journey to learn how to play on a record. You have to learn how control this instrument which is basically meant not to be controlled. You’ve got metal picks and a metal bar on metal strings, across a metal apparatus that amplifies that metal contact, so the opportunity for noise is unbelievable. The good players have spent their life trying to contain it.”

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he says. “With a lot of the recording I do, ”I have numerous slides made of I’m looking for character and that doesn’t different materials,” says Messer, “and necessarily mean a good microphone. If I’m there are a few that produce less string going hi‑fi, I might well use an sE Rupert noise and which I might use sometimes Neve condenser or something like that. when recording. Some players on electric But if I’m going for character, there are guitars like roundwound strings. I don’t. no rules. I’ve used all kinds of things at all I like to use the same set of strings for the kinds of distance from the guitar. whole album. I clean them every day and “For example, on my album Lucky Charms look after them, because I don’t want to there’s a track called ‘Knife Song’. That was have to put a new set on and play them in. recorded using — and this is an extreme Some modern dobro players will change version of what I’m saying — a very cheap, strings every time they play — literally very old AKG microphone that I had in once an hour — because the modern the bottom of a bag. It was so old and sound is that bright, fresh, punchy sound.” on‑the‑edge that the body was live. The Although it’s largely up to the player to whole thing was microphonic! We put it on minimise unwanted noise, being aware a stand about six feet from the guitar and of its sources — principally the contact recorded it as an overdub and it sounded between the fingerpicks and strings at amazing. Why? Because it just did. one end of the guitar, and between the slide and the strings at the other — is important. Of course, for blues slide playing in particular, some of this noise may in fact be desirable, adding some grit and authenticity to the proceedings. Again, the right approach will very much depend on Apply now for an online bachelor’s degree in Music the desired outcome. Production or Electronic Music Production & Sound

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Mic Choice

This is something very evident in Michael Messer’s approach to mic selection. While he favours a single‑mic approach, the mic in question could be just about anything. “It depends if I’m looking for a high‑fidelity recording or a character recording,”

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“I like going for character in that way,” Messer continues. “A lot of the records that I listen to, and a lot of the recordings that I admire, are amazing recordings not necessarily done on amazing equipment. Chess Records, things like that. Go back even further to 78rpm records and the way they would have recorded Robert Johnson. Yes, they used good equipment for the time but with a very narrow frequency range. One of the things we all love about those old recordings is, in fact, that you can’t hear it all properly. “Let’s put it this way — Robert Johnson would sound equally amazing if you were in the room with him, but if he was close‑miked with a hi‑fi sound, I’m not sure it would sound as good as it does the way we hear it. Because it creates a warmth. You’re not getting any highs and lows, it all sits in a bandwidth that’s very comfortable. It works. Modern recording is too good, in a way. Sometimes high‑fidelity recording, like looking in a perfect mirror, is not necessarily the best way of getting a great sound. “If I was recording a solo national guitar for a high‑fidelity recording sitting on its own in the middle of a pair of speakers, then I might use two microphones — perhaps a ribbon at the back of the cone and a condenser towards the bass end. But there are all sorts of problems related to high‑fidelity recording and slide guitar — and that problem is the sound of the slide. I’ve often done sessions where the first thing the engineer or producer says, if they’re not used to recording the instrument, is, ‘Oh God, all that string noise!’. You’re actually better off sometimes putting an SM57 on it — something that just naturally cuts some of that stuff out,

but still gives you a great sound.” In Jimmy Heffernan’s world, where high‑fidelity is more usually the order of the day, mic selection is generally more orthodox. While some Nashville producers, like Bil VornDick, favour a pair of large‑diaphragm condensers such as the AKG C12, Neumann U67 or U47 for dobro, set up in an X‑Y configuration above the instrument (which, let’s remember, sits flat on the player’s lap), Heffernan has particular affection for a classic small‑diaphragm condenser, and he is not alone. “My favourite mic is a Neumann KM84 or KM184,” he says. “One of the guys who is lauded as having some of the best tone ever was a friend of mine named Mike Auldridge who recorded everything with a KM84. I’ve done sessions with him and he would just move that thing around in whatever space he was in and find the sweet spot. Because every space is a little different. He would sit down then move the instrument, move the mic. He was very careful about that. I tend to agree with him. “I get an incredibly warm sound with the KM184. The thing would be to get it pointing in‑between the cover plate [which covers the resonator cone] and the port, not at the centre of the coverplate and the hands. It’s a killer mic. That’s my go‑to. I’ll also use large‑diaphragm mics like the Peluso P12 or Neumann U87. I use a Lawson L47 — that’s a U47 clone — and that gives me great results also. Also to some extent ribbon mics, like the Royer R121 and the Beyer 160. They’re all going to give you good results.”

Mixing & Processing Heffernan also has some specific advice to offer when it comes to mixing the sound

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of the dobro, though he explains that the unusual sound profile of the instrument tends to do some of the work for you. “The frequencies and just the general character of the sound is so different to other instruments that it immediately separates itself from somebody, say, strumming an acoustic,” he says. “Even though this is basically a set of acoustic guitar strings — tuned to different intervals but in the same basic range — you can still sit above an acoustic. Even though you’re not louder per se, you’re cutting through the mix. “I don’t know about everybody, but I tend to thin out acoustics a bit, and that also creates a space that dobros can fill up with the mid range and low end of the instrument’s sound. In fact I will tend to notch that area out on the acoustic guitar precisely so I can create the space for the dobro to live. But it depends. In other scenarios, where I’m up on the higher strings of the dobro, I won’t need to do that at all.” With its singing sustain and wide dynamic range, the dobro certainly boasts some vocal‑like qualities, and Heffernan believes the comparison is an apt one when it comes to dynamic processing. “The one rule of thumb is to treat the dobro like a vocal, especially in a solo context like an intro,” he explains. “You wouldn’t want the

sound of an over‑compressed vocal. You can get away with that on an electric guitar, obviously on a bass, and on certain percussion instruments. But the worst thing you can do is to take the dynamics away from a vocal — or a dobro, which is very vocal‑like. That’s the way I always think about it. It’s like somebody singing. So when I’m dealing with it in a mix or tracking, that’s definitely on my mind. “The thing I hate the most, and I think most players would agree, is too much compression. There’s a lot of dynamic range in the instrument, and often the first thing an engineer wants to do is compress that or limit it. But the true sound of the instrument is supposed to rise and fall so, when you’re mixing, you have to ride it. It’s part of the modern

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style to be very dynamic. Individual notes in a single phrase can be wildly different in volume. A lot of notes I play are going to scare an engineer half to death! Dobro players are called in a lot of the time to overdub, so when you’re tracking, rather than compress or limit, what I would suggest is that you take down the level of all the tracks and try to give the dobro player as much headroom as you can. You’ll sense where he’s going to top out at.” While Messer leans towards emphasising the instrument’s gutsy mid range when recording resonator guitar, especially on characterful blues sessions, Heffernan is more likely to reduce the mid range in preference for the bass and treble in dobro recording. “Sometimes I’ll EQ out around 300 or 250 Hz, just to get a bit of that low‑mid mud out,” says Heffernan. I don’t tend to really do any boosting. I might possibly boost 8‑12 kHz by 1dB depending on the track and the way I was playing on it. If I’m playing very hard, there’s going to be more sizzle in there and I wouldn’t have to do that, but if I’m playing quietly and softly, I will boost up there very subtly. One thing I do like to do is take out 800Hz to some extent. There’s a lot of 800 in here; I kind of look at it like there’s more than enough in there. Subtly slicing out a little of it just helps the instrument open up and blossom. But I’d be very subtle. It’s not heavy‑handed at all, and it depends on the sound of the particular instrument, obviously.”

Shine Of The Times Both players are adamant that their instruments have plenty to contribute

outside of what might be considered standard resophonic territory. “I think there’s a myth around resonator guitars, and the myth is that they’re just slide guitars,” says Michael Messer. “Yes, a guitar with a square‑neck and raised action is only ever going to be used for slide, but anything that’s made with a Spanish‑style, regular guitar neck is a guitar. Just a guitar. And anything can be played on it. “New guitars tend to be set up by the manufacturers for slide players because that’s the majority of the market. Old ones tend to have a high action because older guitars tend to. So lots of players just don’t bother to pick them up because of the way they’re set up. But a single‑cone national‑style guitar is a very powerful acoustic instrument, great for lead playing but also a great rhythm guitar. Used as a rhythm instrument, like electric rhythm guitar, it produces a great sound. It’s powerful — these instruments were made for volume. I’ve used them in that context before, and you wouldn’t really be aware it’s a resonator guitar.” Jimmy Heffernan also likes to use the dobro as more than just a lead instrument. “Away from setting the whole tone of the song around the sound of the dobro,” he says, “it’s incredibly useful in creating parts and lines — ‘sections’ or ‘gang licks’ as we call them in Nashville — where multiple players are playing the same line. If it’s tucked in there it has such a unique quality that it sits in with the guitar or violin or whatever else is playing the line and just gives it so much depth. There are a lot of country records that use that technique. When I produce records I’ll use it quite

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a bit where I don’t play dobro anywhere else in the song. I’ll just play it on the section. You’d have to really know that it’s there to be aware of it.” “Style‑wise,” adds Michael Messer, “I would say that where resonator guitars are pitched in peoples’ minds — blues, country, folk, roots — that’s not necessarily their only place at all. A good example is Mark Knopfler. He uses resonator guitars in ways you wouldn’t necessarily expect. On ‘Romeo and Juliet’, with that melody and that approach to playing the guitar, you wouldn’t expect to find a resonator there. Dave Stewart in the Eurythmics is another example. Further back, on ‘Lola’ by the Kinks, that’s a national guitar starting it off and used as the rhythm guitar throughout. Around the same period, and probably influenced by each other,

was Manfred Mann’s ‘Pretty Flamingo’ — that’s a national guitar all the way through it. “The thing is, it can also be played gently and melodically and sound very sweet,” Messer continues. “Go on YouTube and you’ll find people crashing around playing blues, playing percussively, using resonators for their power and volume. But the sweetness you can get out of a good one is stunning. The sustain, the overtones — no other instrument produces overtones and harmonics like a resonator guitar.”

Thanks To Jimmy Heffernan and Michael Messer. You find out more about them at and Call Of The Blues by Michael Messer’s Mitra is out now on Knife Edge Records.

Speaking Out If resonators introduce some interesting challenges in the studio, this is doubly true when it comes to live‑sound reinforcement; a highly resonant instrument that sounds markedly different depending on where you place the mic is not the average FOH engineer’s idea of fun. With pickup systems getting better all the time, acoustic guitarists are usually happy to simply plug in. But when it comes to resonator guitars, Michael Messer eschews pickups in favour of his own tried‑and‑tested mic technique. “If you sit six feet away from someone playing a National guitar, it’s very different to putting your ear up against it,” he explains. “That’s why I don’t like pickups in them. The best pickup

for a National guitar is the Highlander, which sits in the biscuit under the bridge. It does an amazing job, the best job of everything that’s out there, but it can’t possibly hear the whole picture. You’ve got your ear glued to the bridge, so you can’t hear the whole thing. “Unless I’m forced to plug in for reasons beyond my control, I just use a standard SM57,” he says. “I set the EQ flat, or maybe roll off a little top and bottom so it looks like a frown and not a smile, which is the opposite of what most people would set for an acoustic guitar. And that’s it. I would keep the foldback low, at a similar volume level to my guitar so I’m getting a spread of it rather than being hit in the face with a sound that’s going to cause feedback.

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I then move the guitar around on the microphone — in, out, left and right — to give me different tones and volumes, much like adjusting the controls on an electric guitar.” For dobro players, the picture is slightly different. In the all‑acoustic world of traditional bluegrass, where on‑stage volume is inherently low, it’s possible to use condenser mics on stage, with small‑diaphragm models often favoured. Where the dobro is being used alongside electric bass and drums, or when the player simply wants the freedom to roam around the stage, a clip‑on mic or built‑in pickup is required. The highly rated combination of Fishman’s Nashville Series undersaddle pickup and Jerry Douglas Signature Series Aura processing pedal get Jimmy Heffernan’s nod of approval.

“The undersaddle pickup on its own totally misses the sound of the resonator cone,” he says. “It’s just picking up the strings like a piezo pickup on an acoustic guitar. And that’s what it sounds like — an acoustic guitar, and not a very good‑sounding one. But in combination with the modelling in the Aura pedal, it’s out of the park, it’s unbelievable. “So it’s really the only game in town in terms of dobro pickups, but you’re still going to get feedback issues,” Heffernan continues. “This is a resonant box, you know. It’s built to pick up and amplify resonance and it’s going to take whatever is coming back through the monitors and do that. One option is to cover or block up the ports on the dobro, but really the trick is to have a monitor guy that knows how to EQ the monitor feed to avoid feedback.”

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Cloud Control

In the second part of our guide to Pro Tools’ cloud collaboration features, we learn how to exchange tracks with other musicians. MIKE THORNTON


ast month we started an in-depth look at the new cloud collaboration features introduced by Avid with Pro Tools 12.5. As we saw, something that’s key to these features is that the familiar Pro Tools Session has been complemented by the introduction of a new Project format. Last month’s workshop showed you how to create and share a Project, and this month, we’ll look at the tools that are available for collaborating within Projects.

Tooled Up Global collaboration options are set up using a new Collaboration Tools block, which you can add to the Toolbar section of the Pro Tools Edit window. (When you add a new module to the toolbar it will appear at the right-hand end by default, but if you hold down the Command key on a Mac or Ctrl key on Windows, you can reorder the toolbar modules to suit the way you work.) The Global Collaboration Tools block consists of three buttons. The upwardpointing arrow is ‘Upload All New Changes’; when you have recorded new material or made other changes to existing tracks in the local Project, this upload button will light up to tell you that you have changes that need uploading

to the cloud. Likewise, when one of your collaborators makes changes to any tracks in the Project and uploads them, the downward-pointing ‘Download All New Changes’ button will light up to tell you that there are things that need to be downloaded to bring your Project up to date. The third button in this row of three is ‘Download All Shared Tracks’, and lights up if someone has shared a new track or tracks with you, whereupon you can click on it to download these newly shared tracks. Uploading and downloading can be done in the background, but the time it takes will depend on the speed of your Internet connection. The key thing here is that you decide when to upload or download, when it suits you, rather than updates being pushed into your project when you are not ready. However, if you are happy to have changes sent and received automatically, you can turn on the Auto options under these three buttons. Updates will then be automatically updated so your project is always bang up to date.

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There are three further buttons in the Global Collaboration Tools block, arranged in a column down the left‑hand side. ‘Add Collaborator’ (top) lets you add a collaborator directly within the Edit window; ‘Artist Chat’ (centre) opens up the Artist Chat window; and ‘Notifications’ (bottom) will light up to tell you there is some info relating to this project that you need to check out. There are also track-based collaboration tools that you will need access to. These can be made visible from the View menu, where you will find that Avid have added a Track Collaboration option to the range of things you can view in the tracks in the Edit and Mix windows. These controls appear as a small set of graphical icons; in the Mix window, these are located just above the I/O selector. The leftmost of the three is ‘Track Shared’, and indicates that its track is to be shared in the cloud. When you click on this button, if you then open the Task Manager window you will see all the media relating to that track being uploaded to the cloud. Once everything on that track has been uploaded, the button will turn blue, showing that everything on that track is in the cloud. There are also up and down-pointing arrows, which are upload and download buttons for the individual track. When you make changes to it, you will see that the individual track upload button goes green as well as the Global upload button in the toolbar. If someone else makes changes, the same thing happens with the download buttons.

Le Chat The Artist Chat window is really key to collaborative working in Pro Tools 12.5. It’s where you find people to collaborate with and communicate with them. You can bring up the Artist Chat window either from the Window menu or from the button in the Global Collaboration Tool block in the Edit window, as described above.

“When looking for people you know, don’t forget that the email address they use for their Avid Master account might not be the email address that you use regularly to email them with.” The Contacts section presents two tabs. My Contacts shows people you have already made contact with, perhaps because you’ve already worked together on another project; you can see their screen name and some basic location info, which will show you where in the world they are. The Add Contacts tab is where you can search for and add new contacts. You can search by name or email address, but when looking for people you know, don’t forget that the email address they use for their Avid Master account might not be the email address that you use regularly to email them with. Once you have found whoever you are looking for, you can add him or her as a collaborator and type a message to

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go with a contact invitation. Then they can decide whether they want to link up with you. Below the Contacts section in the left-hand column is a Notifications window, and below that, you can see and access all the Projects that you are involved with. Underneath that, you’ll find the Chat option. All the people you have linked up with will be listed here, so you can select the person you want to chat with and have a text-based chat conversation just like on a number of social media platforms. The difference is that you don’t have to leave Pro Tools to do so. Note that the Chat section is intended for general conversation: if you want to discuss a specific project, use the Projects section of the Chat window. When you select a Project from the

Project section, a window will open, and in the title bar for that project, you’ll see a person icon with a plus sign. This is where you can invite collaborators to a project. Once you click on that icon, a pop-up window opens with a list of all your contacts so you can choose who you want to collaborate with. Once your collaborator has accepted your invitation, your Project will be added to the Projects section of their Artist Chat window, and will be available in the Projects tab on their Dashboard.

Share & Share Alike When your collaborator clicks on your Project in their Artist Chat window, Pro Tools 12.5 will start to open your Project on their system. However, there is a fair chance that when they open your Project

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on their system, it will appear empty, with no tracks other than some rulers. You, as the project owner, need to choose what tracks you want to share with your collaborators: only those tracks will show up in the Project on their system. You can choose to share raw tracks, but to save bandwidth, it may make sense to share just some stems; if you have subgroup aux tracks for the various sections of your mix then you might want to share those subgroup stems, for example. Even though these are not audio tracks, the new Track Commit feature that came out in Pro Tools 12.3 or the Track Freeze feature that Avid

released in Pro Tools 12.4 can be used to ‘print’ audio versions of them that can be shared using the track collaboration tools. Once your collaborator’s system has downloaded all the tracks (they can check progress from the Task Manager window) they can click on the ‘Download All Tracks’ button in the collaboration toolbar module and they will see all the tracks that you have shared with them. Your collaborator can now go ahead and add his or her parts. When he or she is happy, they can click on the ‘Share Track’ button on their tracks and those tracks will be uploaded to the cloud, ready for you to download and integrate into your session.

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Just In Time

Tighten up your timing without sounding robotic, using Logic’s quantisation features. GEOFF SMITTH


n this months Logic technique workshop we look at how to use quantisation to tighten up your recordings. To begin, we’ll look at how to non-destructively quantise MIDI events after recording. We’ll then look at the application of permanent quantisation to tighten up specific note ranges. To finish up, I will show you how to create your own groove templates from a drum sample and apply them to either audio or MIDI.

Non-Destructive Editing Let’s begin by looking at quantising MIDI non-destructively. Record a two bar drum beat that’s based around a 16th note rhythmic feel. If your keyboard playing is anything like mine then your notes will be only roughly in the right places and the feel may be a little looser than you desire. To fix this, double click on the Region you just recorded to ensure that it is selected and the Piano Roll editor is open, then go to the Region section of the Inspector (See Screen 1). The Region parameter area lets you make non-destructive adjustments to a region, which is useful because it means we can return our original at any point. Click on the Quantize parameter and choose 1/16 Note — as we performed our drum beat based on 16th note divisions its sensible to use a similar setting for quantisation. To give your groove more of a swung feel go to the Q-Swing parameter and

increase it from 50 to 60 percent. Do this by clicking the blank space the right of word ‘Q-Swing’, then dragging up to see the parameter value. Notice that as you raise the parameter value Logic takes the second and fourth 16th note of every four and starts to delay them to create the swung feel. You can quickly access a quantise preset with an amount of swing already added by choosing one of the swing presets, for example 1/16 Swing C. A Q-Swing setting of less than 50 percent does the reverse and moves the second and fourth 16th notes earlier in time. If you found the previous quantisation settings too mechanical you can retain an amount of the original feel and groove in your playing by going to the Regions parameter area and clicking the disclosure arrow next to More; this will display all the extra Region parameters. Next, go to the Q-Strength parameter and click and drag downwards, in this case to 90 percent. As you do this, Logic will move the notes 90 percent of the way toward the quantisation setting, leaving 10 percent of the original deviation in place. Decreasing the parameter further will make the quantisation less and less rigid and allow more of the original feel to remain. In Logic you can also set the non-destructive quantisation to be automatically applied to any recorded events. However, the workflow for this is not exactly obvious. Start by clicking in a blank area of the arrange page so that no Regions are selected, and from the Inspector choose

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the quantisation settings you would like to add to your new recording. I find that a quantisation setting of a 16th note and a Strength of 80 percent help to alleviate any timing issues without sounding too mechanical. Now record a new drum beat and this time notice that when you hit stop on the transport your quantisation settings are automatically applied.

A Permanent Fix When trying to improve the timing of a MIDI performance, you may wish to permanently quantise a range of notes from within a region. For example, you may wish to

quantise a single triplet fill that occurs in the middle of an eighth‑note drum beat. To apply quantisation to a range of notes, go to the Piano Roll and find the inspector on the left hand side. Next, select a range of notes with the Pointer tool and from the Time Quantize pop-up choose an appropriate quantise setting (see Screen 2). Alternatively, if your Time Quantize menu is already set to the desired value, clicking on the Q next to it allows you to quickly apply that setting. The Piano Roll also has a dedicated quantisation tool which can be useful if you have lots of small sections within a larger performance that you want to quantise separately. With

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the quantise tool set as the left click tool, draw around a range of notes to select them and then click on one of the selected notes to quantise them to the current Time Quantize setting. To change the quantisation setting at the mouse pointer position simply click and hold, then choose the desired setting from the pop-up menu that appears. Another place you can apply permanent quantisation is from the Event List editor. To do this click the List Editors button or go to View / Show List Editors. Now select a range of notes in the Event List Editor; you can now use the Quantize pop-up menu above the list, and the Q button, in the same way as in the Piano Roll editor.

Groove Templates Next let’s look at how to extract a groove template from your favourite drum sample and apply it to either MIDI or audio regions.

First of all find a recording of a drum beat by a drummer whose feel you really admire. Next, take an extract of their playing and edit it to a bar in length. Ensure the bar of the drumbeat is in time with the sequence tempo. This can easily be accomplished by time stretching/compressing the region to a bar in length: to do this, hold down Option and click and drag the bottom right‑hand corner of the region until it’s exactly in time. We now need to extract the position of the drum events from the drum beat to give us a quantisation template, this is accomplished by detecting the transients. Double-click on the audio file to bring up the Sample editor. Make sure you are on the File tab, then click the Transient Editing Mode button to turn it on, and adjust the sensitivity of the transient detection using the plus and minus buttons next to it until you can see markers for the majority of the

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transients (see Screen 3). If Logic has missed any of the drum hits, for example a quiet hi-hat, then simply use the pencil tool to add a marker. Now, return back to the track that your drum beat is on and go to Edit / Show Flex Pitch/Time to turn Flex Time on. From the Track Header, drop down the Flex Mode menu and select ‘Slicing’. Notice how the region’s parameters area updates to include the Quantize menu. Click in the Quantize drop-down and choose Make Groove Template. Logic will now add a Groove Template to the Quantize menu named after the drum region it was taken from. To apply the groove template to a MIDI region on a software instrument track, record a drum beat on a software instrument track, select the region and then go to the Quantize menu and choose the groove template; again, experiment with the

Strength parameter if the quantisation feels unnatural. To apply your groove template to an audio region on an audio track, record a guitar riff or equivalent, then go to Edit / Show Flex Pitch/Time. In the Audio Track header choose the Polyphonic Flex algorithm and then go to the Regions Parameter area and from the Quantize pop-up menu select your groove template. The audio will be time compressed and expanded according to the groove template. As you can see Logic has a vast array of quantisation options to help you lock your audio and MIDI performances to a particular rhythmic feel — just remember that many of the options are there to help you retain an amount of the feel of the original performance, and that rigid quantisation can be just as fatiguing as overused pitch correction.

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Automation Station Streamline your use of Sonar’s automation with these time-saving tips. CRAIG ANDERTON


utomation is an important part of any DAW — but if you don’t know some shortcuts, editing it can be tedious and sometimes even frustrating. So, let’s consider several tips that streamline the process of using Sonar’s automation. There seems to be quite a bit of confusion surrounding the use of Offset mode instead of standard automation. If you forget you’re in Offset mode, or enter Offset mode unknowingly, it can really screw up your automation. As a result, many people avoid using Offset mode altogether. However it’s an extremely useful option, so it’s important to understand the purpose and uses of Offset mode so you can take full advantage of it.

O Levels The keyboard shortcut ‘O’ used to be the default for selecting Offset mode. However, people would sometimes type ‘O’ accidentally when naming something, go into Offset mode without knowing it, and freak out. So a while ago, Cakewalk removed the default Offset mode shortcut, but I highly recommend re-establishing that keyboard binding. To get the most out of Offset mode, it’s essential to be able to switch back and forth rapidly between it and standard automation. Here’s why Offset mode is so useful. Suppose you’ve used standard automation to create a perfect set of automation moves for a track. However, later on you want to

raise or lower the track level, without altering the automation moves. Type ‘O’ to enter Offset mode, and now the fader essentially becomes a ‘master’ to control the automation (note this works for all automation, not just volume). However, as soon as you’ve entered the appropriate Offset amount, I recommend that you immediately type ‘O’ again to exit Offset mode and return to standard automation. Offset mode should be something you get into, make your level tweak, and then exit quickly. However, if you want to offset the actual automation envelope as it appears in track view (not just add a virtual offset in Offset mode), there’s a way to do that too: 1. Select the Smart Tool and the track with the automation you want to edit. 2. Set the track’s Edit Filter to Automation, then choose the automated parameter you want to offset (we’ll assume for now that automation lanes are hidden and we’re working on the clip itself). 3. Drag the Smart Tool across the section of automation you want to offset in the track itself (or, because the track should still be selected, you can also drag across the timeline to select the automation). 4. Hover the Smart Tool over a clip handle, or over an empty space in the track on the same horizontal plane as the clip handle. The cursor turns into a line with up and down arrows, called the Trim Cursor. 5. Click and drag up to offset the automation upward, or drag down to offset the

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automation downward. Note that when you release the mouse, the automation is deselected to make sure you don’t accidentally vary it any further, so if you want to do more editing you’ll need to re-select the automation. You can also offset multiple envelopes by the same amount: select any of the existing track automation in the Edit Filter, then unfold any automation lanes you want to offset and follow the same procedure as above. With multiple envelopes, you’ll probably find that dragging in the timeline will be the fastest

way to select a region of automation. Note that if automation exists in a lane that is not unfolded, it won’t be edited. You can also edit automation in individual automation lanes. Select the automation in only that lane, hover the cursor just below the top of the lane until it appears as the offset cursor, then drag up or down as described previously.

Down The Lane... The Piano Roll view is controller-friendly if you’re not using a lot of controllers, but with more than a few of them, the strips

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are so short that editing becomes difficult. Fortunately, you can convert MIDI controllers to automation envelopes in MIDI tracks, and edit them as you would audio automation. Doing so also lets you take advantage of editing options like being able to alter curve shapes, or use the Draw tool to create periodic waveforms — sort of like an automation LFO. 1. In the track view, select the clip and region containing the controller to be converted. 2. Choose Clips / Convert MIDI Controllers to Envelopes. 3. In the Convert MIDI to Shapes dialogue box, choose the controller to be converted to an envelope, then click OK. 4. You can continue choosing controllers using the above procedure. If you open up the automation lanes, you’ll now see that your control data has been converted to envelopes; if you close the lanes, you’ll see the envelopes superimposed on the clip itself. However, there is a pitfall. This is a one-way street; once you’ve converted a Piano Roll View MIDI controller to an envelope, you can’t convert it back again to a controller that appears in the PRV.

Copying Automation Suppose you have a stereo effect where you want to use the same automation envelope for two different parameters. Unfortunately, if you copy an automation envelope, create another automation lane, and paste the envelope, it gets copied as the original controller and can’t be reassigned. However, there’s an easy way to do this if you create your original controller changes in the Piano Roll View. For example, suppose the right delay in a synth is Controller 27 and the left delay is Controller 28. First, create the MIDI controller lane for Controller 27 in the PRV, and create your automation curve. Select the automation you want to copy, then create the MIDI Controller lane for Controller 28 in the PRV. Right-click in this automation lane’s header and choose Copy Selected Values to this Lane. You now have two lanes for different controllers with the same data. Then if desired, you can convert these to Track View envelopes using the procedure described earlier.

Clip Service A lot of users dutifully click to create nodes

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in a clip automation envelope, then start moving levels around. But there’s a much easier way to edit clip automation to change gain for a specific region, whether you’re using the Move Tool or Smart Tool. For the Move Tool, set the Edit Filter to Clip Automation / Gain, then select a track. Drag across the timeline to select a region whose gain will be changed in the selected track, then click on the clip’s handle anywhere in the region to be edited, hold the mouse button down, and drag up or down to change Gain. For the Smart Tool, again set the Edit Filter to Clip Automation / Gain, then select a track and drag across the waveform in the clip to select the region to be edited. Move the cursor to just below the upper part of the Clip until the cursor changes to the Trim cursor, and drag up or down.

Quick Switching Often, you’ll want to switch back and forth between a clip and its associated automation; fortunately, there’s an easy to do this. If Clips was selected in the Edit Filter prior to selecting Clip Gain, you can switch

the Edit Filter between the Clip Gain automation and Clips by holding Shift while right-clicking on the clip (if something other than Clips was selected, you’ll switch between that and Clip Gain). And while we’re on the subject of switching-oriented shortcuts, type ‘=’ to switch the filter from Clips to the Volume envelope (again assuming Clips was selected prior to selecting the Volume envelope). Finally, remember that Quick Grouping works with the Edit Filter. Ctrl-click on the Edit Filter, and all track Edit Filters will show whatever you choose. One thing to be aware of is that the Edit Filter truncates long automation parameter names, and the abbreviations may leave you scratching your head as to what is really selected. Fortunately, there’s a simple solution: Hover your mouse over the Edit Filter label, and you’ll see a readout of the full name. Much easier, eh?.

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Making Arrangements We explore some useful functionality in Live’s often overlooked Arrangement page. LEN SASSO


ive’s clip-triggering Session view is what most differentiates it from other DAWs and, as a consequence, Live’s more standard arranging tools don’t always get the attention they deserve. This month I’ll cover two tasks for which these tools are best suited: transcription and collaboration. Both start with timeline‑based material: the song you’re transcribing or the tracks on which you’re collaborating. Session view is very handy for these processes, and I’ll mention it as needed, but the main focus will be the Arrangement view. Because of Live’s split personality, the Arrangement view does have a few idiosyncrasies, which I’ll cover along the way. For more detail on using Live’s two views check out the March 2015 Live column.

Play It Again When learning a song, there’s no substitute for repeated listening section‑by‑section, and that’s a good place to start exploring the Arrangement view because it only involves one or two tracks: an audio track for the song and possibly a MIDI track for recording your playing. Start by configuring a new song accordingly and drag your song file to the start of the audio track in the Arrangement view. For now, make sure Live’s warping for the file is turned off. If you have a rough idea of the song’s tempo, enter it as Live’s tempo. Otherwise,

start Live playing from the beginning of the song and select a segment of the song file in which the beat is clear. Click-drag that segment to a Session view clip slot on the same audio track. Start that Session view clip playing and use the Tap button at the top-left of the Live window to tap in the tempo. The reason this is best done in Session view is that in Arrangement view the tapping causes playback to glitch as Live’s tempo changes, whereas in Session view the tapping has no audible effect. Next select the song file in the Arrangement view and then, in the Clip Viewer’s Sample Box, click both the Warp and Slave buttons to make it Live’s Tempo Master. Manually warping the song file will now produce Song Tempo automation on the Master track — the automation will be grey to indicate that you cannot change it manually. If the song file has been previously warped, you may not need to change any warp markers. If it has not, you’ll see one warp marker at the beginning of the file, and you’ll need to add warp markers to align the sections of the song with Live’s Beat Time ruler. If the song has a rubato introduction, as in Screen 1, play it from the beginning, create a new warp marker at the first beat after the intro and drag it to the closest bar line. Place Arrangement view Locators at the beginning of the song and at the bar line corresponding to the new warp marker, and label them to suit. Start the song playing at the second Locator and

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find the beat that begins the next section. That might be a few bars that you’re trying to learn or a logical song section like a full chorus. Place another warp marker at that beat and then move it until the new section encloses the correct number of bars. Insert an Arrangement Locator there and continue in that fashion until the end of the song file. You can now loop sections between Arrangement view Locators for transcribing or practicing. Screen 1 shows the process for a blues song with a bridge imported directly from my iTunes library. Notice that the time signature changes at the bridge, and thereafter the bar numbers in the Clip Viewer and Arrangement view’s Beat Time ruler don’t match. The reason is that clip view doesn’t support time signature changes. But that doesn’t matter because the song file is the tempo Master and playback is, therefore, unaffected.

Collaboration By collaboration I simply mean starting with a batch of stems (the audio files that comprise an arrangement) and editing that arrangement in Live. The stems might come from the user of another DAW, from a commercial construction kit or remix library, from you using other hardware or software, or from any number of other sources. You might even breathe new life into an old Live project by rendering its tracks as stems and reworking those in a new Live project. (To create stems from a Live project, select ‘All Individual Tracks’ from the top drop-down menu in Live’s Export Audio/Video window.) Here are four tips for when working with stems: • The track display is your highest priority in Arrangement view. To maximize it, close Live’s Browser (Command+Opt+B/

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Control+Alt+B) and Help view (View menu), then hide the I-O and Mixer sections on the right side of the track display. If you have a large number of stems, you might also want to close the Detail view (Cmd+Opt+L/ Ctrl+Alt+L) and invoke Full Screen mode (Cmd+Ctrl+F/F11). • If you have a second monitor, take advantage of Live’s second window (Cmd+Shift+W/Ctrl+Shift+W) to simultaneously see Session view with Live’s graphical mixer. Key commands always go to the window that is on top, but Full Screen mode, the Browser and Help view are only available for the main window. • Stem collections often include effects returns such as reverb and delay. Once you start editing, those stems may no longer accurately represent processing by those effects. Then again, that may

be what you want. • Stem collections usually include a master stem for comparing your changes with the original mix. When you want to do that, assign the master stem’s track to the A side of Live’s Crossfader and assign the Track Group holding the stems to the B side (see Screen 2). In what follows, I’ll assume that all the stems are the same length and in sync. To get started, load each stem at the beginning of its own Arrangement view track. As long as you don’t warp any of the stems, changing Live’s tempo and time signature won’t affect playback; it will simply change the relationship between the stems and Arrangement view’s Beat Time ruler. You can usually use the method described in the previous section to match the Beat Time ruler to the stems by finding a stem in which the rhythm and tempo

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are obvious (typically a percussion track), manually warping that stem (and none of the others), and then setting that stem as the tempo Master. When you need to physically edit a stem, make a copy of the track (Cmd+D/Ctrl+D) and then mute it or use Speaker‑On automation to silence sections as needed. This makes it much easier to revert to the original. In Screen 2 the pink Pad stem was copied and sliced, alternate events were deleted and the remaining events reversed. Speaker‑On automation on the original track eliminates overlaps between the two tracks. Note that you can cut, copy or paste automation without affecting the underlying clip by doing so on a separate automation lane. When you record or overdub MIDI parts in an Arrangement view loop, each pass is recorded as a separate ‘take’ (see Screen 3). When you stop recording, the last take is looped, but you can move the clip’s loop brace to choose the best take. If you

activate Arrangement view’s MIDI Overdub button (next to the Record button), each take is added to the last. If you play nothing during the first take you’ll be able to easily start over. If your computer keyboard is handy, you can even undo (Cmd+Z/Ctrl+Z) during a take without stopping the overdub process. Once you have your stems or transcription master partitioned with Arrangement view Locators, it’s a simple matter to convert sections to Session view Scenes. Right-click the Locator beginning the section, choose ‘Loop to Next Locator’ from the dropdown menu, click the Loop brace and then choose ‘Consolidate Time to New Scene’ from the Create menu. When you return to Session view, you’ll see the new Scene holding clips spanning just the looped material. You can now play and alter those clips as well as record new material on other tracks. Master track tempo automation will apply unless you override it, which you can do by right-clicking on Live’s tempo. You can then enter the tempo in the Scene name to have it set automatically when you trigger the Scene.

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Making Moves Studio One offers the full gamut of automation features — but which ones are appropriate to your needs? LARRY THE O


ast month I took a summary look at Studio One’s grouping facilities and how to apply them. I kind of liked the way that turned out, so this month I’m taking a similar look at level automation in Studio One. As with grouping, Studio One has a full set of level automation functions, and the trick is to understand them and how to bring them to bear on a situation at hand.

Fade To Grey Non-destructive fades, of the sort you get in Studio One and most DAWs, can be thought of as a basic kind of level automation, especially when you consider

long fade times. For example, if one instrument in a mix sustains a note just a bit longer than the rest, shortening the last note on that instrument is easy, but making it sound natural is tougher. Using a longer fade and paying close attention to the curve shape can produce an acceptable decay. Similarly, long crossfades can create smooth transitions between sustained sounds. Crossfades are generally symmetrical in Studio One, but long crossfades often need asymmetrical shapes to sound good. Screen 1 shows a simple workaround: create fades on the two events separately, then overlap them. You can even separate overlapping events, make a fade on each, and then overlap them again; each event will retain the fade shape you set for it.

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Event-level Automation Moving up the hierarchy of level automation features we have the ability to scale the level of an entire event. Again, this is non-destructive, and it’s easy to do: just select one or more events and drag the handle in the middle of the level line, as shown in Screen 2. This is a powerful feature I use a lot, so I’m going to zoom in on it a little. On a broad scale, event-level adjustments are great for matching levels across a phrase. For example, when a solo gets louder and more aggressive as it goes on, I’ll break the solo into phrases and adjust the event levels to get things pretty close, then use other automation methods for finer adjustments that may be needed. Event-level automation is also useful when overdubbing, if the levels of different overdubs vary, or when pieces being edited into a comp have very different levels. The same technique can be used for more small-scale editing tasks, too. Close-miking voices and instruments generally exaggerates their dynamics, and event-level automation on a word-by-word or note-by-note basis can be an effective way to even out rogue notes that jump in level. I have

become quite adept at identifying and selecting words in a vocal that jump out, separating them into separate events by double-clicking or using the Opt/Alt+X shortcut, and changing their event levels to match. I even automate individual consonants at times! Two hints about this method of automation. First, when the automated level difference between events is only a few dB — say 4dB or fewer — it will rarely be clearly audible, and even when it is, it will typically only be so to the trained ear (such as yours). When you get to a larger level differential, however, it does become audible and frequently sounds weird. Those are trickier cases that, alas, I can’t diagnose in this column. Second, there are times when level automation just sounds wrong and a compressor or de-esser is the better tool for the job, even though they have their own sound. It takes close listening to decide on the best approach, but if you try level automation on this scale you may find it more effective than you previously had thought it could be.

Channel & Bus Automation Fader automation is what most people think of when automation is mentioned. One important feature of this sort

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of channel level automation is that it either can be ‘drawn’ with a mouse or performed on a control surface. The latter option offers several attractive qualities. First, there is just the creative satisfaction of performing, of making physical gestures on a tactile controller. Second, well-performed level automation is an excellent way to get the levels of a track pretty well in the ballpark for an entire song in just a couple of passes. Third, it is not difficult to automate a number of channels in a single pass. Fourth, it is great for riding fade-outs by ear. Studio One has the usual modes for performed automation: read, touch, latch, write — there’s no performance trim mode,

but automated VCA faders can provide that functionality. Automating with the mouse offers other advantages. For example, it is difficult to perform fader automation with great precision, or to perform extremely fast moves. Maybe the bass player’s action was too low and there are clicks from the string hitting the pickup every time he whacks a note. Surgical automation to the rescue! A few clicks, a couple of breakpoints, and the clicks are brought low, if not actually removed. Mouse-edited automation can be very fast for setting levels of sections that don’t break cleanly along event boundaries. If the drums get louder from

Automating Instruments Level automation can be applied to virtual and external instruments, as well as to audio tracks. At the note level, pencil tools — including curves — can be used to automate velocity values. Velocity data nearly always modifies volume, but it frequently modifies tonal parameters as well, making it not always suitable for level automation. A few instruments can use polyphonic aftertouch (PAT) for individual note level control, and the emerging Multidimensional Polyphonic Expression (MPE) spec for MIDI provides even more options. To my knowledge, Studio One doesn’t handle PAT at this time, but, to be honest, I don’t have any PAT controllers or instruments to test this with! Virtual instruments reliably have a master level setting of some sort that

can be automated, and they often will have other level settings in their signal path(s), which can include oscillator levels, mix controls, filter output levels and so forth. For example, in Screen 3 you can see that PreSonus’s Presence XT instrument has a volume parameter, and a master gain parameter which is being controlled by MIDI continuous controller 7 (volume) messages. External instruments will nearly always use CC7 for master level control.

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the second chorus until the end of the song, with the ‘smart’ arrow cursor I can select the relevant time range on the drum subgroup’s automation track, then hover the mouse near the top of the track to make the horizontal bar cursor appear. Dragging up or down with this cursor separates the selected range of level automation and moves it up or down as I drag. As I have written before, Studio One’s pencil tool offers parabola, straight line automation, LFO waveforms, and reshaping with the Transform tool, in addition to freehand drawing. These tools can be used to generate or edit audio or instrument automation. Next we come to automation of groups, where we encounter once again the distinction between bus channels and ‘VCA’ faders that we touched on last month. To briefly recap in this context, bus automation is great for applying automation to a mix of tracks, and especially useful for submixes, such as the three or four channels I usually use to record guitar. VCA fader automation makes it easy to reduce or increase the levels of channels going to a submix, or to have control over a group of channels that are not mixed and, in fact, may be routed entirely differently from each other. As already pointed out, VCA faders can be used to trim channel automation.

Choose Your Weapons In deciding which automation method to employ in solving a given need, consider the granularity involved: how small are the pieces of audio you need to automate? If they are very small — let’s call them ‘subatomic’ — you will need to use mouse editing at the channel or group level. If they are a little larger, (say, ‘atomic’) you may find event-level editing effective. Larger durations of audio (er, ‘molecular’?) benefit from performed automation, or the pencil tool applied to larger selections. With all of these tools at one’s command, it is common to end up with several kinds of automation overlaid on top of each other. This can be fine as long as you keep a handle on what you are doing, but if you don’t, it can lead to terrible tail-chasing as you try to find the source of some automation move you don’t like. Studio One, like other DAWs, provides a large toolbox, and often multiple ways of accomplishing a given task. The trick to making best use of this embarrassment of riches is not in using all of the tools that are there, nor in using only a few of them all of the time, but in knowing the spectrum of available tools and considering each one’s usefulness in meeting a particular need.

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Parallel Universe

With Cubase, you can implement an innovative parallel compression technique first conceived by Andrew Scheps. JOHN WALDEN


e’ve discussed parallel compression a number of times previously, but if you need an introduction, read Hugh Robjohns’ excellent article in SOS February 2013 ( latest‑squeeze). Parallel compression can be implemented in different ways to achieve slightly different things, though, and this month I’ll discuss how you can implement producer Andrew Scheps’ ‘rear bus’ technique, which derives from his time working on a Neve console that was designed for quadraphonic sound. Scheps would use the console’s ‘rear bus’ — a second stereo bus that was intended for the rear speakers in a quad system — as a sort of ‘global’ parallel compressor bus. He would often send multiple instruments to a single compressor placed across this ‘rear’ bus, rather than treat each source to its own parallel processing. Using a single ‘master’ parallel compression bus in this way yields

different results from those achieved by setting up multiple parallel compressors. The main difference is down to the interaction between the multiple sources you’ve fed to the parallel compressor, and how the compressor then reacts to whatever element happens to be loudest at any point. In the right mix, and set at the right balance with the uncompressed output, this interaction can create some interesting, almost ‘automated ducking’ effects. Combined with more conventional parallel compression, this can give a mix a certain sense of movement, as elements ‘pop’ gently in and out of the mix. A second benefit is that a single parallel compressor bus is used to accommodate a significant number of the overall mix elements; compared to a more conventional approach, you therefore end up with fewer parallel compressor busses to keep track of, and don’t need to use so many compressors.

The Bus Route So, if you fancy trying this technique, what’s the best way to set it up in Cubase? The starting point is simple: create an FX Channel with a compressor in one of its insert slots. The stock Cubase Compressor plug‑in works well enough for initial demonstration purposes, but feel free to experiment if you have access to something more ‘esoteric’. (We’ll come back to the limitations of Cubase’s

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compressors later.) In most parallel compression contexts, a low threshold (so gain reduction is almost much always being applied), fast attack, medium release and a 3:1 ratio make for a decent starting point, though more (much more) extreme ratios can be dialled in; you need to let your ears be the judge as you balance the compressed and uncompressed signals. Next, create a send from each track you want to be part of the parallel compression processes, almost as if the compressor were an artist for whom you’re creating a bespoke foldback mix. Scheps was reportedly always willing to experiment here, but let’s get the basics sorted before you try anything too sophisticated. A good ‘all in’ approach is to send everything except the drums (or the drums and bass), as this both focuses the compression more on the mid range and will allow your overheads and cymbals to ‘breathe’. A quick way to do this is to select the chosen channels and use the Quick Link function to set up a send from them all simultaneously; be sure to disengage Quick Link when you’re done. In configuring the sends from each track, there are three things you need to ensure. First, leave all the send levels at zero gain (the default value). Second,

ensure that the sends are all post‑fader (also the default). You can check this in the Mix Console, but you’ll need the Channel Settings panel for something else in a minute, so it makes sense to use that here; you can toggle between pre‑ and post‑fader settings by clicking the appropriate button as you hover your mouse over the send label. The combination of these first two settings ensures that the send level from each track is dependent on the channel fader itself, so send levels to the compressor will match the main fader balance. If you’re using Cubase Pro, rather than one of the ‘lite’ versions, you can configure the third setting in the Channel Settings panel. (Don’t worry if you don’t have Pro, as while this stage is useful it’s entirely optional.) From the drop‑down menu in the top strip, toggle the Link Panners option on for each track. When you next adjust the main pan of the track, the pan of the send will be linked to it, rather than being left at the default centre position. This has two useful consequences. First, it ensures that the pan position of the track is matched in the stereo image of the parallel compressor channel. Second, as I’ll explain later,

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it means you can experiment with independent compression of the left and right channels of the parallel compressor.

In The Blender As with any parallel compression approach, it then becomes a simple task of blending the parallel compression bus in to taste alongside your master stereo bus. There are no hard‑and‑fast rules but a good starting point is to gradually raise the parallel compressor’s fader until its contribution sounds obvious, and then back it off slightly.

“It’s an effect which might not be appropriate for every mix... but when it works, it’s really rather cool.” What’s interesting about this approach is the way in which the various sounds contributing to the bus compete with each other. It can result in some interesting changes in their respective volumes and, while perhaps more noticeable in a busier mix with lots of different sound sources feeding the compressor, the end result is not unlike lots of small automation moves having been added, introducing some really nice micro‑dynamic variations to an otherwise static‑sounding mix. It’s an effect which might not be appropriate for every mix, but when it works, it’s really rather cool — so don’t write it off if it doesn’t work the first time you try it!

can route your parallel compression bus to the master. This would replicate the signal routing used by Andrew Scheps, even though, in our case, all the audio is eventually being passed to a single stereo output. However, lots of DAW/sequencer users (myself included) often place a plug‑in or three on their master bus, which means your parallel compressor bus will receive a further (often unwanted) stage of processing, as it’s blended with master output. The workaround is to configure a Group Channel — a ‘master, not‑parallel-compressed bus’, for want of a snappier name — and to route the output of every track to that, rather than directly to the main stereo output bus. Any processing you want to add to the main (not parallel-compressed) mix can be applied here. Both the ‘unparallel bus’ and the parallel compression bus are then routed to the main stereo output channel, where their signals are summed without further processing. It sounds more fiddly than it is!

Along Came Two Busses Providing you mix without any plug‑ins on your main stereo master bus, you July 2016 / w w w . s o u n d o n s o u n d . c o m


There’s More! A number of variations could be built around this ‘rear bus’ approach, but one interesting one is to compress the left and right channels of the parallel compressor bus independently —in other words, use a dual-mono compressor rather than a stereo‑linked one. You still use identical compressor settings on each channel, but this way you prevent a loud sound that’s panned hard right (for example) from ducking a quieter one that’s panned hard left. And once again, treating the left/right channels independently can introduce some interesting small‑scale movement. If you’re to avoid complex routing scenarios in Cubase’s Channel Settings window, this requires that you have a compressor plug‑in that supports a dual-mono configuration, and unless I’ve missed something, that is something none of the stock Cubase compressor plug‑ins offer! The easiest workaround

is to find a compressor that does offer independent left/right channel processing. In the screenshot shown here, I used the freeware Melda Production MCompressor. While working as a stereo or mono compressor, it also offers the option to just process either the left channel (allowing the right to pass uncompressed) or visa‑versa. With two instances of MCompressor inserted on the parallel compression bus, one set to compress the right channel only and the other set to process the left channel only, you are then ready to experiment. (If you’d prefer a single plug‑in for dual‑mono compression, you could check out the VladG’s more colourful‑sounding freebie Molot.)

Louder Or Better? Like all parallel compression approaches, this method has to come with a health warning: as you blend in that overly compressed parallel buss, your mix will become louder, and our ears are easily fooled into thinking louder is better — even when it isn’t! So, before you shout ‘result!’, plenty of A/B comparison is required to check whether the contribution from the parallel compression is really helping the mix rather than just adding level. And if in doubt, you can use Cubase’s in‑built LUFS meter to make sure you’re comparing like with like.

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ABC ‘The Look Of Love’

Photo: Sheila Rock/REX/Shutterstock July 2016 / w w w . s o u n d o n s o u n d . c o m


1982’s ‘The Look Of Love’ paired an ambitious band with an ambitious producer, and the result was a perfect piece of pop music. TOM DOYLE


ew tracks evoke the spirit of 1982 as vividly as ABC’s ‘The Look Of Love’. In an era of post‑punk synth‑poppers, the Sheffield group — guided by high‑concept producer Trevor Horn — managed to create a widescreen sound that blended Nile Rodgers’ and Bernard Edwards’ immaculate productions for Chic with Nelson Riddle’s dramatic orchestrations for Frank Sinatra. “It was like Chic with Bob Dylan over it,” Trevor Horn says today of ‘The Look Of Love’. “That was kind of how they had it in their heads. The guys in ABC used

Artist: ABC Track: ‘The Look Of Love’ Label: Neutron Records Released: 1982 Producer: Trevor Horn Engineer: Gary Langan

to go to a club up in Sheffield, and they wanted to make a record that they could play in the club. If you think about it, in the ‘70s, disco production was the thing that pushed everything forward. I mean, rock production was pretty cool too... But the way that people made dance records was so different. Y’know, they would separate all the drums. The drums would have to be very dry and thick and all that kind of thing.” For Horn, ABC’s musical vision completely mirrored his own at the time, namely to create precise, imaginative, state‑of‑the‑art productions. “I was trying to make something better than you would normally make,” he laughs. “I suppose I was trying to make records to compete with America.” When Trevor Horn and ABC first met in 1981, in a rehearsal room in Queensway, West London, following the band’s debut hit with the Steve Brown‑produced ‘Tears Are Not Enough’, it was clear that they shared certain interests and ambitions. “I was one of about 15 producers they saw,” Horn remembers. “My late wife [Jill Sinclair], who was my manager at the time, had said to me, ‘I’ve found a band who’re perfect for you... this group ABC. They’re intelligent, the songwriting’s really good, but they need something more.’ “So I met them and [laughs] I was probably pretty full of myself. They were definitely! We got on really well. They were into interesting old magazines and I was also into magazines. So they pulled out a couple and I can’t remember what their magazines were, but I had a couple of

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American wrestling magazines. At the time — and y’know you didn’t have the Internet back then — to me magazines were like an insight into different cultures. I think we were kind of kindred spirits in a certain way. “I just remember they said to me, ‘If you get to produce us, you’ll be the most fashionable producer in the world, because we’re the most fashionable band in the world...’”

Irritable Drummer Syndrome ABC evolved from a Sheffield synth group called Vice Versa and, by the time Trevor Horn met them, had settled on a line‑up comprising singer Martin Fry, guitarist Mark White, saxophonist Stephen Singleton,

drummer David Palmer and bassist Mark Lickley. Horn, meanwhile, had been building a renown as both a musician and producer — as the face of the Buggles and their 1979 number‑one single ‘Video Killed The Radio Star’; and the singer in a reconfigured Yes, the production of whose 1980 album Drama he was heavily involved in. Having been a session musician in the late ’70s, he had worked with disco singer Tina Charles and her producer Biddu, with the latter’s recordings firing Horn’s imagination when it came to his own studio work. “Back then, there was no tuition in these things,” he points out. “To become


a record producer was very, very difficult. In order to produce a record, you had to go in a studio and a studio cost a lot of money. When you first got into a studio, you didn’t have a clue what you were doing. So it was tricky. I was doing a lot of work for publishers, demo songs and things like that. And I suppose the thing I learned from Biddu I only learnt because I was living with Tina Charles. “One night she came home and she had the backing track for [1976 number one] ‘I Love To Love’. I’d never heard a backing track before done by a hit producer and it was fascinating because it was kind of brilliant. It was dead simple, there wasn’t a spare thing on it. I must have listened to that about 30 times. It made me realise that I’d gotta stop fucking about — I’ve

got to make the drummer play the right pattern. Because, y’know, back then drummers would get well shirty if you tried to make them play four on the floor.” In 1977, to get around the problem of working with irritable drummers, Horn went into a studio to create a multi‑purpose rhythm track that he could adapt for various recordings. “Y’know, I made myself a ‘drum machine’,” he says, “but I made it on 24 track tape. I recorded everything separately, just a very basic beat — eighths on the hi‑hat, off beat on the snare, four on the floor, done to a metronome with a very grumbly drummer who I had to pay over the odds to do it. And then you could make a drum track by speeding it up or slowing it down to your tempo, and then editing it, and printing it off to half‑inch

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tape and then copying it back onto a two‑inch tape. I used that for making demos and it was a good drum sound.” By ’82, Horn had also had the experience, through making Drama with Yes, of working with 48 track. “I’d sort of ended up producing that along with the other guys [and Eddie Offord]. If you mixed with two machines, it was incredibly slow. So I had a bit of an idea of what I was doing technically.”

Tracing All of this was to serve Trevor Horn very well when he produced a series of hits for pop duo Dollar before moving onto ABC. Their first record together was the 1982 single ‘Poison Arrow’. “We routined the song in a rehearsal room,” he recalls. “I remember going through it and I said, ‘You need something like a middle eight.’ And Martin came up with this bit in the middle where he says, ‘I thought you loved me, but it seems you don’t care’, and [guesting on the subsequent record, Karen Clayton] says, ‘I care enough to know I can never love you.’ I was quite impressed with that. It sounded to me like the kind of idea I used to have that people didn’t want to do. I thought at the time, ‘This is a good sign.’” Entering RAK Studios near London’s Regent’s Park, ABC ran through ‘Poison Arrow’ as a live band take. “I said to them, ‘Is this what you want?’” Horn remembers. “‘Is this kind of good enough? Is this what you saw? ‘Cause we could get it better than this.’ And they said, ‘How do we get it better?’ I said, ‘Well what we’ll do is I’ll program into my 808 drum machine exactly what you’ve played on the drums, Dave. And I’ll sequence the bass [using

a Roland sequencer] on this Minimoog. And then it’ll be like tracing. You play the drums on top of it and we try and get you to play exactly in time with those drums.’ So that’s what I did. “It took about eight hours to program and then I had to leave, ‘cause it was a Friday. I left the engineer recording the drums and then we started work the following week. And the drums sounded great — he’d really nailed it — and then the bass player played along with the bass.” ABC were clearly taken with this method since, when it came time to record follow‑up ‘The Look Of Love’, drummer David Palmer had bought a LinnDrum machine and programmed the beat for the track before the band went into the studio. “He’d got the idea,” says Horn. “You’re talking about 1982 and drummers could play with clicks, but playing exactly in time with another drum track, or as close as you could manage, was a lot harder. Dave was really good and he just got better and better. He was way better than he ought to have been, if you know what I mean, coming from a little band from Sheffield who’d been playing synthesizers that they’d carried around in carrier bags. “With ‘The Look Of Love’, they’d done most of the work when I got to it — they had the Minimoog bass part, they had most of the song. The part that I got involved in was making more of a journey out of it.”

Crack Squad For his work with ABC, Trevor Horn assembled a production team around him who would add much to the band’s sound. First of all, he employed keyboard

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player/string arranger Anne Dudley, whose soaring orchestrations for the group first featured on ‘The Look Of Love’. “I’d met her doing a gig playing with this band down in Wimbledon,” Horn remembers. “Anne came in to do a dep one night and she was fucking blinding. She didn’t just read the music, she played loudly and confidently and she brought the whole thing to life. I said to her, ‘I’m a producer and one day you’re gonna play keyboards on my stuff.’ Of course she probably thought I was a chancer. “She came and played on the last two Dollar records and she was brilliant, particularly on ‘Give Me Back My Heart’. So when I started on ABC I knew I was gonna bring Anne in and all the counter melodies that she put on ‘Poison Arrow’ were terrific. When it came to ‘The Look Of Love’, I told them I thought we should use a string section and they were up for it.” Another key member of the team was engineer Gary Langan. “Gary was brilliant,” says Horn. “He’d tell me off and say, [sarcastically] ‘Are there enough overdubs

on this track? How the fuck am I gonna mix this?’ He’d do things like put an echo on something and I’d go, ‘My God! Can we get away with that?’ I mean, he and I, we got crazier and crazier. The vinyl cutting engineer was always moaning at us.” The final addition to the production crew was JJ Jeczalik, brought in to program the Fairlight CMI sampler that Horn had recently purchased for an at‑the‑time eye‑watering £18,000. “Yeah, you could buy a house for that,” Horn points out. “But don’t forget I’d sold like 10 million copies of ‘Video Killed The Radio Star’, so I could afford it. And I knew what it could do. “I think Kate Bush, Geoff Downes [the Buggles/Asia] and Peter Gabriel had them, so I had possibly the fourth. But you see, everyone else was kind of serious with it, where I was having a laugh really. I always remember, just before I bought the Fairlight, the Synclavier people phoned me up and the guy said, ‘Y’know, if you’re buying a sampling instrument, the Fairlight would be alright for gimmicks, but our

Fairlight Remix For the 12‑inch ‘US Remix’ of ‘The Look Of Love’, Trevor Horn utilised the Fairlight to create a groundbreaking extended mix which chimed with early hip-hop records, using the sampler to bend and ‘scratch’ various elements of the track. “I actually think that was the best 12‑inch I ever did,” he says. “Everything kind of came together on that one. What I’d said to JJ was, ‘I want bits of everything on the Fairlight — bits of the strings, bits of the vocal, just anything.’ I seemed to manage to get a narrative through it, playing these

things on the Fairlight keyboard. “Gary was buttoning stuff in and out from the rhythm track to give me some sort of bed to play it over and I was just winging it and doing loads of tries at it. We’d always play it loud, and then get that onto the half‑inch tape. It goes ‘goodbye goodbye goodbye’ and I just went mad with things. But it was exciting, and it just seemed to get better and better and then we spent ages editing it from the half‑inch into a sort of good shape.”

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machine is like a scientific instrument.’ I said, ‘Well I’m going for the gimmicks [laughs].’ And I did buy a Synclavier a few years later, but I never quite got out of it what I got out of that Fairlight. “I almost immediately gave it to JJ, ‘cause he just devoted his life to it for a while. And because he was a non‑musician, he would always approach it from an unusual angle. His idea would be to put a tennis match over something, not an oboe solo or whatever. That was much more fun.” The tracking sessions for ‘The Look Of Love’ were done at Good Earth Studios in Soho, while the mixing was completed at Sarm East Studios near Brick Lane, using one of the first SSL 4000E consoles in

the country. “‘The Look Of Love’ was the first track that I mixed with a computer, because we’d just got the software for the SSL,” Horn recalls. “Gary Langan and [co‑engineer] Julian Mendelsohn said, ‘We want to use the computer’, and I was like, ‘Fuck the computer, I hate the computer.’ I’d tried using [Neve automation system] NECAM and it had really screwed me up. So they pacified me and said, ‘Give us, like, five hours.’” On ‘The Look Of Love’, there was much in the way of ear‑grabbing sonic trickery, not least Stephen Singleton’s saxophone parts swathed in AMS digital delay and Lexicon 224 digital reverb. “I said to Gary, ‘You’ve got to get a sound on this record where all


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that Steve has to do is play one note and it sounds fantastic.’ And he did it. I thought it was amazing.” In a great example of Horn’s eye for conceptual detail, meanwhile, he insisted that the female voice saying “Goodbye” in the second verse of ‘The Look Of Love’ was recorded by Martin Fry’s former girlfriend who had dumped him. “I did suggest it,” he laughs, “because if you’re gonna do things like that, I just think you’ve got to kind of make them real.”

The Lexicon Of Love From here, Trevor Horn, ABC and the rest of the team embarked on the making of the band’s debut album The Lexicon Of Love, chiefly recorded and mixed at Sarm East. Controversially, however, before work began, Horn suggested that the band replace bassist Mark Lickley with Brad Lang. “I talked them into getting a better bass player, which maybe wasn’t a kind thing to do,” admits the producer. “I thought that they needed someone more like Bernard Edwards on the bass, and that it would slow the whole process down enormously if we didn’t have that. It wasn’t very fair on the guy who’d been in the band, and in fact it lost me the U2 gig. Bass players in other bands would kind of be a bit scared of me. But Brad Lang was quite brilliant. He was a combination of Bernard Edwards and Jaco Pastorius. When I asked him to come along and play with them, they got it straight away.” By the time of the album’s recording, Horn remembers, post his sequenced parts ‘tracing’ method on ‘Poison Arrow’, ABC were, along with Anne Dudley, tight enough to lay down most of the backing

tracks live. “I don’t remember labouring like hell over anything,” says Horn. “By the time we had Brad Lang and Anne, we had a pretty hot little band. Something like ‘Valentine’s Day’, that’s a live take.” After a satisfactory take had been achieved, Anne Dudley would get to work overdubbing layers of keyboards. “I was big into proper keyboards,” Horn says, “like the Wurlitzer and Fender Rhodes and grand piano. One of the key sounds on ‘All Of My Heart’ is a Fender Rhodes doubling a grand piano all the way through. The great thing about working with Anne is if she comes up with an idea and you say, ‘I like that,’ she’ll stop and she writes it out very quickly. Other people I’ve worked with — and I’ve worked with some brilliant keyboard players — can occasionally forget what they just did [laughs]. So the piano part that she played on ‘All Of My Heart’ was pretty well worked out and she was capable of playing it exactly the same each time.” Horn heightened the drama of show‑stopping ballad ‘All Of My Heart’ by suggesting the dropdown that comes before the concluding line of the chorus, while on the Motown‑ish soul stomper ‘Many Happy Returns’ he edited the multitrack to expand the outro around Dudley’s improvised Fender Rhodes part. “She just doodled around for a second and I loved it,” he says. “I edited the tape and made the whole section twice as long. The band were away at the time and I just did it. It was quite a complex thing to do because I had to make a copy of the multi track, edit that new section in, and then make it work and play over all the joins. And as I was doing it, the voice at the back of my head was saying, ‘If they don’t

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fucking like this, it’ll be a real problem.’ But they liked it and I thought, Well, that’s great. I can do this shit and nobody’s gonna complain.” For Mark White’s guitar parts, the sounds varied from clean and funky to soaked in the latest digital studio effects. “We worked a lot on Mark’s guitar sounds,” says the producer. The [chiming, echoing] one at the start of ‘Date Stamp’ is lovely. We just hung everything on it we could.”

Strings Attached The string sessions for The Lexicon Of Love, working to Anne Dudley’s arrangements, were done at both Abbey Road Studios and Advision Studios in London. Burned by a bad experience in the past, however, Trevor Horn admits that he took a back seat during this part of the process. “The last time I’d done strings had been

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in 1978 on a Bruce Woolley record,” he says, “Our string arranger wasn’t very good and it wasn’t very well written out. But the string players were also a little bit bloody‑minded and they literally stopped in the middle of a take as the clock went through one. I was furious. I threw the cheque on the floor in front of the string players, I was so angry with them. “So when we were doing the strings [on The Lexicon Of Love], I just stayed in the control room and chain‑smoked dope [laughs]. It meant that the highs were high and the lows were low! But Anne did a great job of the arrangements. She seemed to understand from quite early on how to write the parts so that the string players could understand what she wanted. It’s getting the expression right, because you’ve got to remember, the string players, they’ve all got to feel it together. If you don’t spot the great take, and you drive them into the ground, they’ll become

dispirited just like anybody else.” Recording Martin Fry’s vocals, Horn — of course a singer himself — tried to make the process as painless as possible for the frontman. “In the Buggles, God bless his cotton socks, Geoff Downes would make me sing one line for two hours, and I found that so depressing that all my joy at singing would go. Martin was still finding himself back then, so because I knew how to use 48 track, I gave him like seven tracks and just got him to sing the song seven times. And, y’know, guided him a bit, but took the pressure off that whole dropping in thing where you’ve got to sing the same word over and over again. I put the vocal together from the tracks and it made vocal sessions better for him. He didn’t dread them.”

Take Two... The mixing of the elaborately‑arranged The Lexicon Of Love was, perhaps




understandably, quite tricky. In the end Trevor Horn and Gary Langan were forced to mix it twice. “We mixed and then we cut it,” Horn remembers, “and then I played the cut back and I didn’t like it. I asked for another two weeks. I said, ‘I wanna go back, I know I can get it better.’ It was too soggy‑sounding. I wanted to make it more crisp‑sounding. So we went back and we just worked harder. Pushed it harder, kind of thing, and were a bit more ruthless. I think the last track we mixed was ‘All Of My Heart’ and the computer broke down halfway through it. I threw a fit, and I said, ‘We’re gonna mix it without the computer.’ We tried that for about five minutes and I realised we had to get the computer working again.” Horn recalls that the final mix of ‘All Of My Heart’ was completed around midnight on a Sunday, but that when saxophonist Stephen Singleton came in to hear it, he noticed a glaring sonic error the others had missed. “He said, ‘Sounds great, I love the mix, but at the end there’s a funny noise on my sax.’ So we played the multitrack back and we couldn’t hear it. And then we played the half‑inch of the mix, and sure enough he was right. There was a little wow. There was a bearing gone in the half‑inch machine. “So we tried to get another half‑inch machine at two o’clock in the morning. I remember phoning Abbey Road and the guy goes, ‘Sorry mate, there’s nobody here.’ I couldn’t get any other studio on the phone and there was another session in at Sarm East at nine in the morning — Pete Collins who produced Nik Kershaw. My wife managed to postpone his session, took him out for lunch and everything, while we managed to get another half‑inch and print the mix.”

The mixes on The Lexicon Of Love proved to be as dramatic and dynamic as its songs, with much in the way of volume contrast, and the strings and key drum fills pushed to the foreground. “That’s what we were aiming for,” says Horn. “We were aiming to keep you interested all the way through, not let it flag anywhere. Keep the arrangements exciting, because that’s what Nelson Riddle used to do with Frank Sinatra.”

The Lexicon Of Love II 2016 sees Martin Fry, the sole remaining member of ABC, release The Lexicon Of Love II, working with Anne Dudley and producer Gary Stevenson. Asked recently why he hadn’t involved Horn in the production of the sequel, Fry said, “It would have been wonderful to work with Trevor Horn again, but it would have been a bit like having George Lucas and JJ Abrams in the same room.” “I’m not disappointed in the slightest,” Horn states. “It would be the kind of thing that either I would be totally involved in it or not involved at all. So I’m totally not involved in it.” As to why both ‘The Look Of Love’ and The Lexicon Of Love have endured as classics, the producer has a simple theory. “You listen to them and you can feel the era,” he says. “And also Martin’s lyrics are just the best. In that genre I’ve not really heard many people say something meaningful and interesting. Let’s face it, dance lyrics were kind of “love to love you baby” and “boogie wonderland”. And you’re talking about lines like “a sunken ship with a rich cargo” [from ‘Show Me’] on a dance record. It was absolutely unique and it caught a lot of people’s imaginations.”

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Secrets Of The Mix Engineers: Matt Hyde The success of Deftones’ Gore is reward for the band’s restless experimentation — and for the work of engineer, producer and mixer Matt Hyde. PAUL TINGEN


eftones’ musical openness and penchant for experimentation have earned the band the moniker “the Radiohead of metal”. Their music moves dramatically between a wide range of moods, sonic colours and musical styles. The band’s eighth studio

album Gore is their most experimental to date, yet also their highest-charting, reaching number two in the US, five in the UK and topping the charts in Australia. The album’s critical and commercial success obviously also has to do with the quality of the songs, the playing, the arrangements and the production. Singer and guitarist Chino Moreno and the

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album’s engineer, mixer and producer, Matt Hyde, explain how they and the band as a whole managed to conjure up an album that’s inventive, cohesive and successful.

Metal & More Matt Hyde’s track record is as multi-faceted as the music of the Deftones, in that he is also a musician, playing guitar and keyboards, and has worked with everything from hip-hop (Cypress Hill) to US roots music (Jonny Lang), to all variations of rock, from stoner to alt to hard (Monster Magnet, No Doubt, Sum 41, Porno For Pyros). He does, however, have a penchant


Written by Chino Moreno, Stephen Carpenter, Frank Delgado, Abe Cunningham and Sergio Vega. Produced by Matt Hyde and Deftones.

Photo: Frank Maddocks

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for metal music, having helped steer bands like Slayer and Hatebreed to great success. Hyde first worked with Deftones as an engineer on their previous album, Koi No Yokan (2012), which was produced by Nick Raskulinecz. Towards the end of the project, Hyde also got involved in the production side and, according to Moreno, he and the band developed a “great rapport”. For this reason, the return call in 2014 didn’t come out of the blue. “I was contacted in the Spring of 2014 with the message that Deftones might be interested in working with me again,” remembers Hyde, “but the first thing that happened was that Stephen [Carpenter], the guitar player, hooked up with me in the Summer because he wanted to work on his guitar tones. During the making of the previous record we had spent a lot of time with his Fractal Audio Axe-FX unit on modelling some of his amps, as well as mine. So we went into the studio for a week and a half in the Summer of 2014 and we both brought every amp that we own — that was a lot of amplifiers! — and modelled many more sounds in the Fractal. A little while later Stephen called me saying that the band wanted me to co-produce a new record with them.”

Old Ways Deftones were keen to continue working in the same way as they had on their previous albums: in time-honoured, money-consuming, 20th Century rock & roll fashion, first writing and arranging songs in a rehearsal space, and then recording and mixing them in a commercial studio. Hyde was happy to oblige, and in the Autumn of 2014, set up with the band in a rehearsal space in Los Angeles.

“They had done some demos at the beginning of 2014, not with the whole band, but just preliminary stuff done by some of the guys. We started by going through that material first, and figuring out what was cool, and they then began writing new songs as a collective in the room. We would get together a couple of weeks at a time each month, and I would record all the rehearsals using an Apogee Duet with my laptop and Pro Tools 10. They were playing in the room with a PA — it was like a production rehearsal before a tour — and I would take a stereo feed from their monitoring mixing board. “Jamming together in a room is how they have always written. They don’t sit in front of computers and write stuff. It’s not the way modern bands and kids work these days. This was old-school: a band jamming until they find a part that feels good, and then combine that with another part that we liked. It was a slow process, but during each two-week period we probably got between three and four songs and cool parts and ideas together, and we developed the latter during the next period. We also figured out a lot of the atmospheric stuff that ended up on the record. Chino will have melody ideas the whole time, but usually he writes the words after the track has been written. “In general terms, the feel and dynamics are really critical with this band, as they do the loud-soft thing better than anybody, so they worked a lot on that. My role was to be like a facilitator. They write, and bounce stuff off me. I helped them a little bit sometimes in steering things in certain directions, and helping them solve problems with melodies or harmonies and assembling arrangements and organising

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their ideas. But I was not the guy coming up with the ideas. These guys know what they are doing!”

Sparks Fly Moreno’s recollections tie in with those of Hyde, though sometimes with different emphasis. “We [the Deftones members] have known each other since we were kids, and because we are so close to each other, what the others are going to do can be a little expected. For this reason we try not to have preconceived ideas of what we are going to do for an album before we get together. Instead we like to go into songwriting situations with nobody knowing what to expect, and this adds to the excitement. “The sparks of uncertainty and of being caught off guard by someone doing something unexpected are the highlights of making music together. Many of the best songs and ideas come from instinct. When somebody presents you with a new idea, your first reaction to that is going to be the most organic. So there was a lot of free-form stuff going on during the rehearsals, even though, like Matt said, maybe a third of the record came from a riff that somebody might have recorded on his own, or played for us in the room. But there never was anything that was really fleshed out as a complete idea.” Logistics also played a crucial part in the proceedings, with the band members living in different parts of the US. Drummer Abe Cunningham, bassist Sergio Vega and Moreno made the best of their stay at a hotel close to the rehearsal space, at some very un-rock & roll hours. “The three of us would meet up for coffee every morning at 8am,” explains Moreno, “and we’d discuss what we’d one the previous day, and what we were going to work on that day. The rehearsals did not start until noon, so we had four hours to kill every day, and because we all live in separate cities we don’t get much time to be together, and we wanted to use the time we had as productively as possible. Sometimes we’d go to somebody’s hotel room, and hash out ideas, playing electric guitars without amplifiers. But nothing really took shape until the five of us were in a room at the July 2016 / w w w . s o u n d o n s o u n d . c o m


rehearsals. As much as we can have ideas apart from each other, nothing really takes shape until everybody reacts to what it.” One example was the album opener, and first single, ‘Prayer/Triangles’, which, recalls Moreno, “was one of the last songs we wrote. Sergio and Abe were just jamming, with Sergio playing that bass line, and I came up with that little delayed guitar line in response, which formed the core of what became the chorus. Stephen [Carpenter] was running late, and he came in while we were playing, picked up the guitar and immediately launched right into the chorus chord progression, with Abe playing double-time snare. It almost sounded like U2 or something, not like Deftones at all. There was no talking, the song was written in literally one hour. These are the special moments that are the essence of the band. Although the song now sounds like Deftones, it does not

sound like something we’ve done before. It was something fresh that came into being without us even thinking about it. It was one of the most organic experiences you can have being in a band.” There’s been a lot of speculation that the many aspects of Deftones’ music can be traced back to specific band members, with guitarist Stephen Carpenter seen as the metal-head and Moreno bringing in influences from outside metal, and so on. Moreno, however, plays these differences down, stating, “I know we’re seen that way, but it’s not that black and white. People assume Steph is the metal guy, Sergio is the alt-rock guy, I’m the new wave guy, whatever. But honestly, as metal as Stephen is, he writes a lot of stuff that is not metal. The initial idea for ‘Phantom Bride’, for example, came from him. He was playing that guitar riff over and over, and I suggested we

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make a song out of it because it’s very infectious. Steph reacts to what we do in ways that even I would not expect. So yes, we do all have our core personality traits, but when we make a record, we don’t stay in our corner. Everybody gets in each other’s territory, and that is what makes it such a collaborative experience.”

Watt’s Up In February 2015, after a writing and arranging period that had, on and off, lasted several months, the band and Hyde reconvened at Megawatt Recording for the actual recordings and the final mix. Megawatt is a top-class recording studio that has existed since 1978, when it was called Lighthouse, and saw legendary stars like Chicago, Earth, Wind & Fire, and John Fogerty within its walls. Renamed Bay 7 Studios 20 years later, the facility was visited by the likes of the Beach Boys, Tom Petty and Santana. It was reincarnated as Megawatt in 2012, and consists of Studio A, a tracking room with a vintage Neve 8058 Mark II desk plus two 12-channel Neve sidecars and a 95-square-metre main recording area, and Studio B, a mix room featuring an SSL 4080G+ desk. It was a comparatively old-school project for Hyde, who has his own DAW home studio where, he says, “I can make an entire record, and where I also master. It allows you to do great stuff for very little money. I can make a record for $5000 and I can make a record for $500,000!” Gore didn’t cost half a million to make, but spending several months at Megawatt nonetheless involved considerable outlay. “It is what Deftones are used to,” Hyde explained. “They have been around

since the early ’90s, and are comfortable with certain types of technologies and have certain expectations of how things are done, so for them it was a definite decision to use a recording studio and a desk. There also is something special that you get from sending things through a big console that is different from working in the box. To a certain degree you do get what you pay for. There are certain sonic things that happen in a studio environment that are advantageous.” “We chose to record in a commercial studio for a few reasons,” elaborates Moreno. “Number one is obviously quality, though I have heard plenty of records that have been done in the computer, and I can’t tell the difference. It could be more of a placebo kind of thing where it feels more real when you are in a studio. The first time we made a record it was in a real studio, so it is something that we are very used to. The five of us cutting the tracks in a proper studio, with a nice board and with nice outboard, is important for our frame of mind. When you are sitting behind an SSL or a Neve console you feel like you are doing the real thing. And because you’re spending money, it puts pressure on you. You better do your best and really focus! As humans we can easily get lazy if we don’t have any pressure.”

Bumps In The Road By June 2015, the recordings for Gore were complete (see ‘Tracking’ box), and Hyde began mixing the album at Megawatt Studio B. However, there was an unexpected glitch: the band were on tour and not entirely happy with the mixes Hyde sent them. As a result they

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decided to hire another mixer, only to find that this didn’t work out either. Moreno explains: “I like to be there during the mix, or at least that somebody from the band is present, because there always are certain details that you want to hear and make sure are there. When Matt started mixing, I kept missing elements, so we said, ‘Let’s get somebody else to mix it.’ But while these mixes were OK, we were missing many nuances, because that person had not been there during the recordings. In the end we decided to return to working with Matt. He’d had a month off, and came back to the mixes with fresh ears, and we were now able to be there most of the time, and at this stage the mixes came together right away. Matt had been working on the album for a long period, and I think it was good for him to step back and get some perspective. Matt is a very smart individual and very outspoken, who gives feedback right away, which I like, and we clicked really well. With Matt there for the entire project he ended up being like

a sixth band member, and that ended up working really well. He is a big part of this record. His hands are all over it!” Hyde recalls that, for him, the final mix was simply an extension of the recording stage. “Honestly, I feel like I have the vision for a song the whole time we’re working on it. I know already during tracking what I’m really going for, and the roughs are usually pretty close to where I want to be. Many of the vocal and other effects are dialled in during tracking, because the way I worked is that after tracking drums, I went home and I did a mix in the box, and I printed a drum stem, and I then at Megawatt recorded the bass to that drum stem. I would then do a bass stem at my studio, and recorded the guitars to the drum and bass stems, and so on. “In that way I could bring up the sounds that we were after and lay them out over the desk pretty fast, and people had consistent monitor and headphone mixes while tracking. By the time we did vocals I had all the instruments stemmed out, and

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we would cut the vocals to that. The stems may have changed as we were working on a song. The drums stems in particular often evolved during the first few days of tracking, with me repeatedly modifying the stems until they were the way we wanted them. For the final mix all these stems got broken back out again with individual tracks spread out over the board. “I moved to the SSL mix room at Megawatt for the final mixes, which took about a month. This is where it got interesting, because I had done a lot of parallel compression with my stems in the box, and I continued using many of them during the final mixes, but I also had parallel channels on the board. The drums in ‘Prayer/Triangles’, for example, were broken out over 17 individual tracks coming out of Pro Tools, and then there were another 12 channels with analogue effects. I did have somewhat of a mix template for all the songs, but there also always were elements that were particular

to each song, which I usually routed to eight channels on the board that I reserved especially for that purpose. “I mixed the entire record one song at a time. And then all the comments came in. Also, many of the mixes were evolutions. By the seventh song you may have hit on something really cool that you also want to do on the previous songs you mixed. So there were recalls from that perspective as well. We even re-amped and re-tracked certain parts at the last minute! In the modern age everyone wants to recall stuff endlessly. We’re at a point where we can manipulate things to death, especially when you work in the box at home and you have no time limits. One of the nice things of working in a commercial studio is that it does put a limit on things. When I work on a desk I make my mixes as recallable as possible, despite working on an 88-input console, and when the comments were all in, recalled about four to five songs a day.

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To recall the console, get the mix set up including outboard, make the changes, and print the mix took me about one and a half to two hours.”

One Thing At A Time “Generally speaking,” says Hyde, “my mix method is to bring up the drums first, get them to sound the way I want, bring the bass in, then bring the guitars in, and once I have the entire music track going, I bring in the vocal. Then I start messing around with everything in, and make it sound good together. There is a certain amount of value in listening to things in solo, or per group, so you get some knowledge about the frequency content and compression, but the most important thing is how it all sounds together. “As I mentioned, I ran a combination of plug-in effects that I had retained from the in-the-box tracking stems, and desk EQ and compression and outboard. Volume automation also was a combination of in and out of the box. It’s easier to do the really precise stuff that happens only at one moment in the box, but I did all the vocal and guitar rides on the board, using Ultimation. I don’t do many moves on individual channels, but instead do most of my moves on the subgroups in the centre section of the board. It’s about gain-staging stuff. There is only so much you can do in the box. It’s about how you hit compressors, and the desk faders are the final determiners of how things are going to hit the stereo mix bus.”

The Session Within a short duration of 3’38”, ‘Prayers/ Triangles’ is a typical example of the way Deftones works with light and shade, soft July 2016 / w w w . s o u n d o n s o u n d . c o m


and loud, beauty and brutality. It starts with a heavily distorted electric guitar in the right speaker, travelling to the left, and then an atmospheric verse, with what sound like electric guitar harmonics put through a delay, followed by an all-out double-time‑snare chorus with heavily distorted rhythm guitars, that still bears some faraway echoes of U2. The middle eight and outro also are full-on, in essence extensions of the chorus. The song’s Pro Tools session comes in at 115 tracks, breaking down in, from top to bottom, nine drum effect tracks at the top, 30 actual drum tracks, six bass tracks, 10 Stephen Carpenter guitar tracks, 20 Chino Moreno guitar tracks (in the middle of which are two kick drum sample tracks, called ‘Death’ and ‘Rockhead’), four electronics tracks, 15 vocal tracks, 14 vocal effect tracks and finally three mix tracks.

One thing that’s striking is the amount of editing in the individual drum tracks. Some of the wave form regions look black, which echoes the way Mutt Lange treated some tracks on Muse’s Drones album (see SOS October 2015), though Hyde insists the editing is fairly minor. “I was editing for time and feel. But these are live drums, plus some samples I took from the sounds of his actual drum kit, mostly from the kit with fewer mics.” • Drums: Audio Ease Altiverb; Kush Audio UBK1 & Clariphonic; Waves SSL E-Channel; SoundToys Decapitator; Valley People Dynamite; dbx 160; Peavey Kosmos; SSL bus compressor; ITI stereo EQ. “The nine drum effect tracks are at the top, because it’s the way I organise things at my house when I do mix stems. The first four are different Altiverb reverbs:

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Cello Studios Room 1, AMS non-linear, AMS RMX16 and Lexicon 480L. Then there’s an overall drum bus, with the Kush Audio UBK1 compressor and the Kush Audio Clariphonic EQ, and it has a send to the Cello room. The snare bus has the same plug-in treatments, the overheads and toms buses have the Waves SSL E-Channel, and the room bus the SoundToys Decapitator. Below these nine tracks are four ‘drumpadprocess’ tracks, which have an SSL channel on the inserts and the sends go to a parallel Decapitator track. They are also sent to four outboard Valley People Dynamite compressors, to get some extra pop on the kick and snare. “There’s quite a bit of outboard on the drums. In addition to the Dynamites there’s also a dbx 160 on the kick and snare, and a bus with an interesting unit called the Peavey Kosmos, a sub-harmonic synthesizer. And there’s an analogue stereo drum bus that goes through the SSL compressor and an ITI stereo EQ (the predecessor to the Massenburg). Really, most of what I am doing in the box is a copy of what I used to do on the board. When

I have both available, I just do the same thing twice and make it work together. It may seem overkill, and it’s rather unscientific, but I just listen and turn knobs until it has the right impact and everything sounds the way I want it.” • Bass: Waves Trans-X Multi, Linear Phase EQ & Scheps 1073; Teletronix LA2A; SSL EQ & compressor. “Because Sergio has such a unique bass sound, I ended up using plugins I don’t normally use on bass. His sound doesn’t have the same weight as you normally get, so I had to use a subwoofer cabinet to add some extradeep low end. I then had to contain it, which I did for example by using the Waves Trans-X Multi compressor. There’s also a lot of stuff happening on the board. All the Fractal stuff comes out of one channel and all the SVT stuff out of another channel. They are being EQ’ed, using the Waves Linear Phase EQ, and compressed with the Waves Scheps 1073, and come up on my sub bus, 7-8, on the console, on which I also have the kick. The insert on the bass on the desk has the Teletronix LA2A outboard compressor, and I also used desk EQ and compression.” July 2016 / w w w . s o u n d o n s o u n d . c o m


• Guitars: Waves Scheps 1073 & CLA Guitars; SSL channel; SoundToys EchoBoy; API 550b; Kush Audio Clariphonic. “Steph’s tracks are in red, with two tracks, L and R, for each part. I used the Waves Scheps 1073 to add some colour and brightness, just on the left tracks, and then they go through Aux 1, which has the Waves CLA Guitars plug‑in for compression. There’s a lot of stuff happening out of the box on the guitars, including SSL board EQ and compression. On Chino’s guitar tracks I have several EchoBoy delays, an API 550b and again the Kush Audio Clariphonic. There’s a track called ‘M160dup’ which is me sending the guitar part to an amp to get some special effects for the verses.” • Vocals: Antares Auto-Tune; Waves Renaissance De-esser, Renaissance Vox,

VEQ4 & LoAir; McDSP Filterbank; Crane Song Phoenix II; iZotope Trash; Avid Mod Delay III; SoundToys EchoBoy; AIR Distortion; Audio Ease Altiverb; ADR Compex F760X RS; Empirical Labs Distressor; AMS harmonizer; Yamaha SPX90. “There are many vocal tracks! The main verse vocal track is called ‘New Verse TN01’, just below the ‘Aux 1 dup1’ track, and below it are some harmonies, and the chorus lead vocal is right below the ‘Aux 5’ track, ‘NWC6...’ All vocal tracks first have tuning from Auto-Tune, then a Waves RDe-Esser, and then the Waves RVox for compression. I then send the vocals to several subgroup busses, where they get de-essed and compressed a second time, so I don’t have to process so hard the first time. I find that by touching the vocals lightly

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twice, it sounds more natural than when I hammer them once. Some of the vocal tracks also have the McDSP Filterbank or the Crane Song Phoenix II tapeemulation plug-in, set to Iridescent for some extra sparkle and also to take some harshness away. I love those Phoenix plug-ins, they are really subtle. “HC means hardcore, and ‘HC Vox’ is the track on which Chino is screaming in the bridges and the outro, and has the RDeEsser, RVox and Filterbank, plus the iZotope Trash, using a patch called ‘Smooth Overdrive’. It gets sent to an aux track called ‘HC Sub’, which again has the RDeEsser, RVox, the Waves VEQ4 and again the Phoenix II. “Further down are the vocal aux effects tracks. They are each for one effect, respectively an Avid Mod Delay set to 166ms, an EchoBoy with a non-linear multitap thing, another EchoBoy with a classic 100ms slap, an AIR Distortion, an Altiverb reverb set to a Lexicon 480 gold hall, another filtered long echo, and a Space Echo, the latter two both from the EchoBoy. Aux 3 is a long filtered delay that answers Chino singing “prayers prayers”, and aux 5 has the Waves LoAir sub‑harmonic generator. Most of my delays are filtered. The effects often are so heavy that I want to chop off many frequencies, so they quickly disappear in the mix, like a ghosting effect, otherwise everything would begin to sound garbled. “In terms of outboard, I had the ADR Compex F760X RS on a parallel bus, which is a really fast compressor, ideal for limiting and spanking. It is side-chained to some really extreme EQ. I’m also using the Distressor on the vocals,

again as a parallel. Instead of using the [SoundToys] Little Altar Boy plug-in, I ended up using the outboard AMS harmonizer and also the SPX90 for some pitch changes, all old-school ’90s stuff!” • Stereo mix: Sontec Mastering EQ; SSL Quad compressor. “I printed back into the Pro Tools session via Burl A-D converters. The session was 24-bit/48kHz; 96kHz would have driven me crazy for this project! I’m using a Sontec mastering EQ and an SSL quad mix‑bus compressor on the stereo bus. But I keep my mixes at a standard level. I leave the whole loudness thing up to the mastering engineer, in this case Howie Weinberg. “I actually master 75 percent of my projects, using the iZotope Ozone plug-in, but I prefer to keep the mixing and mastering processes separate. When I send my mixes to the artists and management I crank them up with the Massey L2007, so they get a kind of fake mastering, just for them to have perspective. The L2007 gets it loud enough, and does not damage the mix. I prefer to keep my original mixes dynamic and punchy, and I don’t want to overly destroy them before it gets to mastering. The loudness things is crazy, and this is the best way of dealing with it. Chino and I discussed mastering options, and we were mostly concerned with tone, giving the music warmth and low-end love. I didn’t want that ‘crunchy/ fried’ sound that I hear on a lot of things coming out recently. Howie managed to get it super-loud and still retain the dynamics and warmth. I was really happy with his work.”

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Tracking The Old-school Way “By the time we arrived at Megawatt Studios and started recording, the chord progressions were locked down, the song structures dialled in, and the arrangements complete, even though some arrangement choices came into being at Megawatt at the last minute,” says Matt Hyde. “When we listened to what we had recorded we would sometimes have better ideas for a drum beat, and [keyboardist and turntablist] Frank [Delgado] was always working in the background on his synth sounds and samples, and his parts changed a lot during the recordings. “We laid down some of the stuff live, particularly some of the interlude pieces and intros and outros, but 90 percent of the album was recorded via overdubbing. Typically I’ll start with the rhythm guitar, to get the tuning and

song structure down, but this is such a unique band that we did it in a more traditional, ’70s way, stacking things from the bottom up. So we began by recording the drums, then the bass, then Stephen’s rhythm guitars and after that his lead guitars, then Chino’s guitars, and Frank’s parts were the last to be added during the main tracking sessions. The vocals were recorded at the very end, in late Spring, early Summer 2015, and following that there was some more work on Frank’s electronic stuff, and a few more sessions with Steph for extra lead guitars. “I recorded many of the instruments using the Neve 8058 desk and the two Neve 12-channel sidecars, so I used Neve mic pres, mostly 1073s, and some 1066s and 1089s. Abe had two different drum kits, and I recorded one

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with a close-miked setup with gazillions of mics, and the other with more of an Andy Johns setup, with three mics and a couple of spot mics. I tend to change mics and move them around all the time, but some general mics would remain the same — like on the multi-miked kit a Shure SM57 and Neumann KM84 taped together above the snare, a Sennheiser MD441 underneath the snare, AKG C414s on the toms, and a C12 for overheads. The kit with the three mics had three Sennheiser MD421s, plus an SM57 for the snare top, and a Neumann U47 FET, Sennheiser e602 and NS10 on the kick. Room mics could have been a pair of RCA 44BXs halfway down the room, plus a pair of Violet Amethyst mics and a pair of Coles ribbon mics, and I might also have had a Telefunken ELA M 251 as a mono room mic. “Sergio has a really unique bass sound, and he uses the Fractal Axe-FX and has a SamsAmp Bass Driver DI. I recorded his bass in different ways, from his DI going to a couple of Neve mic pres and various compressors like a Urei 1176 or a Tube-Tech, and on his Rivera sub cabinet I used a Sennheiser e602 mic, going into a Neve 1089 and a Retro Sta-Level. “The guitar cabinets were recorded with a Beyer M160, and at other times an SM57 or a Sennheiser 421, and the room mic was an AEA R84 ribbon. The guitar sounds were a mixture of Rivera amplifiers and stuff coming straight out of the Fractal, using Palmer DI boxes, in different combinations for each song. In some cases I re‑amped his guitar again later on. Steph plays an eight-string guitar, and with Sergio playing a six-string bass there’s a lot of

overlapping going on, which makes it harder to mix, but also is unique. It’s part of the band trying to break a few rules. “With Chino’s guitar it is a little different. He mainly uses just one amplifier, the Rivera Knucklehead Tre, which has certain sounds that he loves. To record his vocals I used a Bock Audio 241 microphone, going into a Martech MSS-10 mic pre, and then a dbx 160XT, which works well on his voice. At other times, particularly for the distorted stuff, he’s holding an SM58 going into the MSS-10 and a Maag EQ4 500-series. He likes to play with effects while he’s singing, so he also had the Eventide Mixing Link which allowed him to put effect pedals in his vocal effects chain. He operated the Mixing Link with his iPad, and we’d print that, and we’d also print his dry signal. We recorded his final vocals at a small studio in Oregon in the middle of nowhere, which was perfect because it allowed us to get away from everything and just focus on the vocals. I brought my laptop, a Universal Audio Apollo 8P Thunderbolt interface and the vocal chains I just mentioned. “I recorded Frank’s stuff DI. We spent quite a bit of time at my studio working on sounds. He uses a Nord, which is his main hardware synth, and he works in Ableton a lot, and I brought in some different soft synths that I like, for example Omnisphere and a Mellotron plug-in called M-Tron Pro. He also likes to use guitar pedals on the synth sounds, making use of a Little Labs PCP Distro to insert guitar pedals in the chain, like the Electro-Harmonix Memory Toy and Holy Stain, and so on. He definitely likes to mix analogue and digital.”

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INTERVIEW Photo: Sophie Gransard

Film Director & Composer John Carpenter is not only a cult movie-maker, but also a pioneering electronic composer. John, his son Cody and collaborators Alan Howarth and Daniel Davies explain how it’s all possible. PAUL TINGEN


he path from the music world to the film world is well trodden. David Bowie, Madonna, Sting, Alanis Morissette, Björk, Bob Dylan, Debbie Harry, Cher, Courtney Love and Tom Waits are just a few of the many stars who have tried acting, while David Byrne, Dylan, Madonna and Rob Zombie are among the musicians who have directed feature movies. Moves in the opposite direction, however, have often been less successful. Steve Martin

is accomplished at bluegrass banjo and Jack Black is active in Tenacious D, but the musical exploits of William Shatner, Gwyneth Paltrow, Bruce Willis and Eddy Murphy have all strained the boundaries of artistic credibility. Legendary film director John Carpenter is breaking the mould in many respects here, enjoying a career change in his late ’60s to become a bona fide rock & roll star, with critical acclaim to boot. Carpenter has directed classic horror, action and sci-fi movies like Dark Star (1974), Halloween

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(1978), The Fog (1980), Escape From New York (1981), The Thing (1982), Big Trouble In Little China (1986), Prince Of Darkness (1987), Village Of The Damned (1995) and more. And like Charlie Chaplin and Clint Eastwood, to name just two, Carpenter is famous for also scoring the music to many of his own movies. In Carpenter’s case, there is a twist, in that he’s a pioneer of using synthesizers, with his scores in many cases having become almost as famous as the movies themselves. Most people will be familiar, with the high arpeggios, five-to-the-floor stomp and atmospheric four-note motif of the Halloween movie theme, to give just one example, where all the elements work perfectly together to conjure up a suitably spooky atmosphere. Carpenter’s film scores, in a number of cases made together with composer Alan Howarth, have proved so outstanding that they have laid the seeds for an entire electronic music sub‑genre, synthwave (also known as retrowave), which emerged in the mid-2000s, and also quotes Vangelis and Tangerine Dream as its influences. The last two movies Carpenter directed, Ghosts Of Mars (2001) and The Ward (2010), were both commercial and critical flops, which goes some way towards explaining the man’s latter-day career shift. His new direction became apparent in 2015, with the release of his first non-soundtrack album, Lost Themes. Like his film scores it is dominated by synth sounds and arpeggios and mixes spine-chilling moods with rock influences. Made together with his son, Cody Carpenter, and Godson, Daniel Davies (the son of the Kinks’ Dave Davies), Lost Themes reintroduced Carpenter to the

world of glowing reviews, with one critic noting that the album “evokes his past without rehashing it, delivering a complete and immensely satisfying portrait of his music along the way”. One year on there’s the follow-up, Lost Themes II, as well as something even more essential to an aspiring rock star: a world tour. At the time of writing 30 dates were already scheduled, beginning in late May in the US and ending in November in Europe, taking in three UK dates in late October, in Manchester and London, with Carpenter appropriately headlining ‘Release The Bats’, All Tomorrow’s Parties’ Halloween extravaganza. Further in line with the film-music-world crossover theme, the two Carpenters and Davies will be accompanied for this tour by Tenacious D’s backing band, meaning that audiences will be treated to a genuine rock & roll line-up with drums, bass, two electric guitars, and the two Carpenters on synths. A YouTube video of the opening track from Lost Themes II, ‘Distant Dream’, showcases the six-piece sounding eerily like a prog‑rock band.

The Quiet Life “Making music is far more intimate and less stressful than sitting on a film set, let me tell you!” laughs John Carpenter. “It’s a lot less stressful. It’s all about the music, just the three of us making music in a room, or on tour the six of us cranking away, which is just joyous. Listen, I am 68 years old, and all this has happened to me in the last couple of years. I’m just a lucky dog, you know. It’s just great! It has come late in my life, and it’s just wonderful.” Neither his association with music, nor his predilection for using synths, nor his shift from film- to album-making, came by

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Photo: Phil d’Angelo

design. His father was a music professor, composer, and a concert violinist, who introduced his son to classical music and started him on the violin. The latter, recalled the 68-year, “was fine, apart from one big problem: I had no talent on the violin! It is one of the most difficult instruments to learn, and I was never any good at all on it. So I rebelled, and moved on to piano, and then guitar, and bass guitar, and rock & roll. And later on at film school I began scoring my own movies because there was no budget. “Despite having very limited means, I still wanted my scores to sound as big as possible, and using synthesizers was one way of doing that. I realised that if you multitracked yourself you could build up

a sound that’s big and orchestral. That was the main reason for using synths. I also just love the sound of synths, because it is so unique. Of course, my classical background helped me, as well as watching lots of movies and being in love with classic movie music written by composers like Dimitri Tiomkin and Bernard Herrmann. I was also influenced by a synth group called Tangerine Dream, who did a great soundtrack for a movie called Sorcerer [1978], and by Goblin [the Italian prog rock band who became famous for their soundtracks for the horror movies Profondo Rosso (1975) and Suspiria (1977)].”

The Theme’s The Thing Most synth enthusiasts are tech-heads,

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ranging from the nerdy to the obsessive, but Carpenter claims almost complete ignorance on the technical front. “I don’t remember what synths I used. My big thing was that I never bought any equipment until quite late in my career. Part of the gig was that the people that I worked with had the equipment, so I’d go into their places and use their equipment, with them engineering. This was how Alan Howarth and I worked on the soundtracks we did together.” Alan Howarth was born in New Jersey and moved to Ohio during his high-school years. He set his first steps on the musical path playing bass guitar in bands. He was keyboard tech for Weather Report’s Joe Zawinul during 1976-1980, and went on to work as a sound designer on the first six

Star Trek movies, as well as Raiders Of The Lost Ark, Back To The Future 2 and 3 and many more films, and as a composer has contributed to the scores of several movies. Howarth is promoting something he calls Natural Resonance (naturalresonance. net), which states that 440Hz is an arbitrary frequency standard for A, and that instead frequencies like 432, 427 and 421 Hz have a natural resonance with specific energy centres in the human body and are therefore more attuned to the wellbeing of our mind and body. Howarth is currently working with Magic Leap’s Sonic Arts team on their revolutionary approach to “mixed reality”. Alan Howarth and Carpenter worked together for an amazing ultra-productive eight‑year period, which Howarth calls

Going Live At the time the interviews for this article took place, John and Cody Carpenter and Daniel Davies were deep in rehearsal for their coming tour, which will be a novel experience for all of them. As mentioned in the main article, the three additional musicians are the backing band to Jack Black’s Tenacious D: John Spiker on bass, John Konesky on guitar and Scott Seiver on drums. The Carpenters will be playing keyboards, and Davies guitar and occasionally keyboards. Davies explains that his live rig will consist of a non-reverse Gibson Firebird, Gibson Les Paul, Fender Jazzmaster, a “white, handwired” Vox AC30, and a Dave Smith Mopho X4 and Roland System 1 synths, plus a laptop. Cody Carpenter will be using a laptop with Mainstage software and a MIDI

controller, and the Yamaha MX49 keyboard, though he remarks that they may “just be renting keyboards when in Europe. My dad will be bringing just a laptop and an M-Audio MIDI keyboard. The drummer will be triggering some of the arpeggios from a Roland trigger thing, but other than that we’ll mostly be playing the songs live, so we have to arrange them a little differently for the band. But I’m really excited about this tour. It’s crazy.” John Carpenter: “Mainstage allows us to reproduce the synths sounds that we have. We’ll be playing a mixture of music from the Lost Themes albums and my film scores. Most people who come to the concerts will want to hear these scores. Yes, there’ll be visuals as well, but I’m not going to tell you what. You’ll have to come and see!”

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“one of the greatest experiences in my lifetime”. Carpenter shot 10 movies in this time, all of which were scored by Carpenter and Howart, starting with Escape From New York (1981) and ending with They Live (1988). Howarth vividly recalls Carpenter’s lack of interest in technical aspects of music-making. “John did not want to know anything about it. He was like, ‘I don’t need to know that stuff, that’s your job.’ I was the gearhead and I had the gear, so he came to my studio at my house, and my job was to keep the synths in tune and up and running and make sure the red light was on. John just sat in front of the black and white keyboard keys and pushed them. “He’s a master of themes, particularly simple themes that communicated the mood and idea of his movies. Most of the scores we did were just him coming in and starting to play, with me being sort of an electronic music producer. I created sound

palettes for him, and kept putting these sonic tapestries in front of him which he called ‘my electronic colouring books’, and I ran the sequencers and kept all the technology going. It was a really magical time to be working with him. He used to say that he really enjoyed doing the music for his movies, that it was when he was on vacation. He could leave all the pressures of being a director behind him when he came over to my place, turn off the phone, and just be a composer and relax.”

Just A Phase Fortunately, Alan Howarth does have clear recollections of the gear the two of them were using. “What I call ‘phase 1’ was from the moment John and I started working together on Escape From New York in 1981 up until Christine [1983]. My studio was called Pi West, and the initial bank of keyboards there when

Photo: Alan Howarth

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Photo: Alan Howarth

John and I started out working together were the Sequential Circuits Prophet 5 Rev3, a Prophet 10, an early ARP Quadra and two ARP Avatars — one Avatar was integrated with a Sequential Circuits

Model 700 programmer — an ARP Sequencer, an Oberheim Four Voice, and an Emulator 1 sampler. I also had the Linn LM1 drum machine, which drove our main arpeggiator, the ARP Sequencer, using

Dave & Daniel Davies The Kinks’ Dave Davies and his family, including the young Daniel, moved from the UK to LA in 1992. They settled in the same street as the Carpenters, and the teenage Daniel even spent a couple of years living with them. Davies junior started out playing guitar, occasionally asking his famous father for some advice, but mostly being self-taught. After several years playing in bands he now works at his home studio in LA writing songs and soundtracks, using a setup consisting of Logic, an Apogee Duet, Adam A7X monitors, and a small collection of BAE preamps, a dbx compressor and some mics like a Shure SM7 and an AKG C414.

Davies is not as enamoured with working in the box as John Carpenter, stating that he’s “still trying to figure out a good marriage between soft and hardware synths. It gets a little boring just being in the computer all the time. I want to turn knobs as well. Things are so clean in a computer that old synths sound rather noisy when you plug them in, but the old stuff also introduces character and unpredictability, which is great. And the endless options in soft synths can be overwhelming. I’d rather use a few simple things than one thing that can do everything. I have thousands of sounds at my disposal, but I probably use just 20 or 30 as my starting points.”

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clock pulses. I later got an Emulator 2, which became one of our main pieces and had a very musical arpeggiator on its own. I was very much into programming our own sounds, and Dave Smith of Sequential Circuits really liked the original sounds I had made for Joe Zawinul and asked whether he could use them in the first Prophet 5 presets. So many of the Prophet 5 presets were my sound designs, especially for brass, strings and stuff like that. “I initially had a Tascam 80-8 half-inch eight-track tape recorder at Pi West, but we quickly decided that we needed more tracks, so I got a Stephens 24-track, the reason being that it’s somewhat portable, and there were some serious steps leading up to where I lived, so getting an Ampex up there was somewhat prohibitive! Two-inch 24-track tape was our recording

medium. I expanded the tracks a little bit with the half-inch eight-track, using SMPTE and also the Audio Kinetics Q-Lock sync system, so I could lock both tape recorders to video. John loved that, because it was the first time he’d been able to play while watching the movie! Prior to that he’d been playing to a click and a stop watch, and the music was synchronised to the images later on 35mm mag stock at the movie studio. “Phase 2 began with Big Trouble In Little China. I had a new house and built a studio in the garage out back, which I called Electric Melody Studios, and where I had an Ampex MM1100 24-track instead of the Stephens. I added a Kurzweil 250 synth and a Prophet VS, both of which became main pieces, and also got the Linn Drum LM2 and an E‑mu SP12 Drumulator. I started getting into computer-based sequencers

Photo: Alan Howarth

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as well. The first one I got was MOTU Performer on the Mac Plus. But the crown jewel which we got in 1986 was a Synclavier system. It was so expensive that I bought it on payments, it was like buying a house! I decided to go for the Synclavier rather than the Fairlight because it had added a Macintosh as its interface. “Made by New England Digital, the Synclavier was a synthesizer, sampler and digital recorder. The direct-to-disk system could be expanded to 16 tracks of hard‑disk recording. Initially it had 64 voices and 64MB of RAM, meaning it was a bit like a gigantic Emulator 2. The direct-to-disk system allowed you to

trigger sounds and disk-based tracks from its keyboard, so RAM was no longer an issue. All this was really cutting-edge for the time! In using RAM and disk-based recording we were already using the tools of the future. Because I had a Synclavier I was a late-comer to the DAW. I did not jump to Logic until 2003. Eventually I phased out the Synclavier. It took 30 Amps just to have it on standby! I now work in both Pro Tools and Logic, but am now looking at PC and Nuendo.”

Heading Into The Box After Carpenter stopped working with Howarth he did, for a while, own some

Photo: Alan Howarth

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Photo: Dominic Mancini

keyboards, in particular, he recalls, “the Korg Trinity, which I used to score a couple of films with, and then the Korg Triton. That was it for a while, until I hit Logic Pro.” The mention of Logic Pro connects Carpenter to the subject of the making of Lost Themes I and II, and by now it should come as no surprise that his switch to a DAW also

wasn’t by design. “My wife got me into it about eight years ago,” recalled Carpenter. “We renovated the house we were living in, and she bought me a computer with Logic Pro. I looked at it, and it terrified me, but my son Cody said, ‘It’s not that tough,’ and started working with me on it. Over

Cody Carpenter Cody Carpenter’s father introduced his son to music and to movies at a very young age (yes, even to the horror movies), and he then went on to play piano and later guitar and drums, and to study music composition at the University of California in Santa Barbara and at the Occidental liberal arts college in LA. Cody contributed to the score for his father’s movies Vampires (1998) and Ghosts Of Mars (2001), and also wrote and performed the soundtracks for two episodes in the Showtime Masters Of Horror TV series, ‘Cigarette Burns’ and

‘Pro-Life’, both of which were directed by his father. Cody Carpenter is active as a musician with his Ludrium solo project, which sometimes features additional musicians. Both Carpenters are also big fans of video games, with Cody quoting the soundtracks of late ’80s and early ’90s video games as a big musical influence, and adding that he’s “also heavily influenced by progressive rock legends like Genesis and the recently departed Keith Emerson. But similar to my dad, I’m more into improvising than coming to music from a theoretical point of view.”

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a period of time I became conversant with it, and now I love it. It’s tremendous. I can’t tell you how great it is. You can have an infinite amount of tracks, and all these plug-ins, and the sounds of the synths are just unbelievable. I recall having to start up tube synths in the old days, giant things that you had to tune. I also had to work with an engineer. But now I can do it all myself at the push of a button. It’s great! Logic is just so user-friendly, it’s fantastic.” Once he felt comfortable enough to work with Logic, John Carpenter, with help from his son Cody and Daniel Davies, would record music himself in his home studio, where he has a typically 21st Century studio setup, with his Logic Pro DAW complemented by Adam A5X monitors, an Apogee Quartet interface, a couple of microphones, and a few MIDI keyboards. Cody spent much of this time in Japan, from where he would communicate with his father and Davies via email. Eventually their collaboration resulted in Lost Themes I, which was successful enough to galvanise the trio to work together over a shorter and more focused period in 2015 to create a second album. “All three of us being in the same room together was more rewarding than the way we worked on the previous album, with Cody being in Japan,” says John. “The three of us usually started with an arpeggio or something like that, or Daniel or Cody would bring in a sketch, and we’d work on that, improvising and expanding it into something bigger. We did it in all sorts of ways, sometimes the three of us playing at the same time, sometimes just two of us, or one. There’s a lot more guitar work on the new album than on the first

one, because I just love rock & roll, and wanted to have more guitars. The lead guitars were played by Daniel, but Cody also played guitar. We all programmed drums. We shared many of the tasks and this is what made it a lot of fun.”

New Gear, Old Feel “One or the three of us would come up with an idea, a riff or something,” elaborates Cody Carpenter. “My dad often likes to start off with an arpeggiated line or sequence, and we’d build on that. For example, my father came up with the bass line for ‘Distant Dream’ and we put a melody on top, and then we created the entire structure for the song. We all played keyboards, working mostly in MIDI, and Daniel and I played guitar and also some bass here and there, and some percussion. One song has a percussion sound that’s a sample of my dad’s voice, heavily effected. Daniel did most of the drum programming because he’s really good at that. There also was a lot of editing in Logic. Some songs have bits from other songs, and we’d edit sections, move sections around inside songs, and so on. Editing was a huge part of the process.” According to Davies, “the guitars were recorded DI, going through a BAE 1073MPL and then a BAE 321A, and inside of Logic we used amp simulators, like Logic’s Amp Designer, or Native Instruments’ Guitar Rig, or Softube’s Vintage Amp Room. If it started sounding too similar, I’d change to something different. The keyboards that we used were my Roland JP8000, John used the Korg Triton a lot, and we also had the Korg MS2000 and Korg Kronos in the room. For the sounds we mostly used soft synths and

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samplers, like Spectatronics’ Omnisphere, which sounds great, and Arturia and U-he stuff, and we also had tons of plug-ins by FabFilter and Softube and others, to change the sounds.” Despite the fact that most of the sounds on Lost Themes I and II are from digital sources, the sonic universe on the two albums has a very analogue feel, with many warm, fat, fuzzy, analogue-like sounds that stand in contrast to, for example, the more glassy, digital sounds Cody Carpenter himself likes to use for his Ludrium solo project. “For sure, my dad and I are really into older, classic synths sounds,” agrees Carpenter junior, “as opposed to the really modern stuff. We definitely tried to go for a more analogue feel. There’s something romantic about the old synths, but in terms of convenience it’s so much easier to just use a soft synth. We also used Native Instruments’ Kontakt, and a plug-in emulation of an Oberheim synth. We’d go through the patches, and if we liked the sound, my dad would turn on the arpeggiator to see what we could do with it. We’re not that sophisticated when it comes to programming sounds, so we tended to go for presets, and then tweak them a bit and put some plug-ins on.”

Just The Music Like all of his music, the Lost Themes albums exemplify John Carpenter’s distaste for what he calls the ‘Micky Mouse’ approach to movie scoring, where the music tries to manipulate the viewer’s emotions almost second-by-second. Rather than try to micro-manage the viewer’s feelings in this way, Carpenter has always allowed the music to follow its own inner logic, and used it to invoke

a strong general atmosphere over longer periods of time. “The combination of music and image and understanding how they work together definitely is a big part of having grown up with my dad,” recalls Cody Carpenter. “And in 2005 my father and I did the music together for one of these Showtime TV series. We went somewhere to record, with an engineer to help us, and we just ran the video and sat down at the keyboards and improvised to that. Then we’d go back and worked on the bits we liked best. We played to the final masters, so didn’t have to go through this process of adapting the music to later versions of the film. But when we were working on Lost Themes, I didn’t have any images in my mind, and I think my dad did either. It’s more a stream-of-consciousness kind of thing: just come up with a musical idea and work on that. We don’t think about moods or ideas, it’s just free-flowing musically.” “When you’re scoring for a movie, you’re servicing and supporting the story and the images,” adds John Carpenter. “The music always came after the images. Last year we did the theme music for the Zoo TV series, and that was back to the old days, because we were scoring to images. But for the Lost Themes albums it was just the sounds, just the music. Sure, the music we were doing would stir up images in my head, but they have nothing to do with the titles. I just made them up, using one word for each of the titles on Lost Themes I, two words for the titles of the songs on Lost Themes II, and I may use three words for the titles on Lost Themes III. Not writing music to visuals was fabulous. Very freeing. It freed me up to making music in ways I had not done before. It’s pure, it’s great, it’s wonderful!”

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SAMPLE LIBRARIES Gothic Instruments Dronar Hybrid Module Kontakt Instrument Dronar is a Kontakt instrument that translates the everyday into the epic. When fed with a simple triad chord, it cunningly fills out the middle range, provides a building‑shaking bass and inserts highs to rival the most discreet online pharmacy. From an inviting interface, vast washes of sound can be animated by modulation, effects and arpeggios and topped off with an ambient sound effect. All that cinematic goodness emanates from source waves that equate to 16GB, although fortunately these are rendered in compressed form, reducing the download to half that amount.

Once installed, Dronar is found in Kontakt’s ‘files’ menu rather than the more accessible ‘libraries’ option. Although extensive tailoring is available, the main panel has a mere half dozen dials with which to sculpt the Lows, Mids, Highs, FX, Movement and Intensity. Probably the first control to try after loading any of the 300 presets is the


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mod wheel. It’s routed to Intensity and is the most direct means of fading in the different layers. The Movement dial is equally useful; it sets the intensity of multiple arpeggiators — arpeggiators that include step‑sequencer‑style modulation of key parameters. The patches are placed in categories such as Alien, Tension, Strange, Horror, Strings, Deep Space, etc, and they serve as brilliant starting points for composers into genres such as horror, crime, fantasy, science fiction and miscellaneous unpleasantness. I’m still trying to forget the terrified cries of the forest toads, apparently menaced by a JCB. And the monstrous but compelling ‘Organ Of God’ wasn’t an encounter I ever expected to


have outside a bad mushroom trip. Only in the pad category are a few lighter moments to be found, suitable for a spot of new‑age dolphin‑fondling, perhaps. Gothic Instruments plan to explore a wider range in future modules (with discounts offered to Dronar owners). While this collection is polished and contemporary, it can sometimes be a challenge to differentiate one vast, brooding backdrop from the next. However, that’s my only complaint as the quality is consistently high. If I had to pick an instant favourite, it would be the strings, which are superb. Top marks should go to the sound designers who have made each creation a near‑perfect thing already, just awaiting chordal input. Speaking of input, less is definitely more and the simplest chords invariably worked best. Dronar expands them in ways that would tax a dozen trained octupi, with a humble triad demanding 54 notes from Kontakt in one example! Continuing with the effort‑free theme, rapid but significant sonic

tweaks are there for the taking, especially when you replace the mouse with a hardware MIDI controller. Until now, I’ve always associated the term ‘atmosphere creator’ with our elderly Patterdale terrier. Dronar is just as difficult to ignore, offering as it does a significant chunk of finely‑produced, often luxurious material at a very reasonable price. Whether you’re into dark drones or questing for the ultimate pad, ambience or tripped‑out cinematic moodgasm, it comes highly recommended. Paul Nagle £59.95

Xtant Audio Model Brass Kontakt Instrument Whether you are interested in the whole orchestra, or just a specific orchestral section, there are plenty of sample library or virtual instrument choices now available at a range of prices. If you are on a more modest budget, but interested in a collection of brass sounds — and perhaps July 2016 / w w w . s o u n d o n s o u n d . c o m


Model Brass something that is suitable for both general classical and jazz duties — then Xtant Audio’s Model Brass might well appeal. The library is designed for use with the full version of Kontakt 5.5, contains around 10,000 24‑bit samples (around 2.6GB of sample data), provides a collection of 15 solo brass instruments — divided into French Horns, Trombones, Trumpets and Tubas — and, because of the design approach, is very playable. That design approach involves a combination of both samples and physical modelling. This means that many of the performance articulation elements normally associated with sample‑based orchestral instruments are actually handled by the physical


modelling and Kontakt scripting. The result is a more compact library and less time key switching between performance styles. Instead, key switching is used to toggle between legato and polyphonic modes and a number of different mutes (plunger, cup, bucket, etc) as appropriate for each instrument. These comprehensive mute options do make Model Brass suitable for jazz as well as orchestral styles. The scripting/modelling approach allows you to use a number of MIDI CC options to add expression, and the custom Kontakt interface includes options for switching/blending between four different microphones for each instrument and some nice ambience options. The design also deals with round‑robin issues and legato playing in a very pleasing (to the ear) fashion and, from a playing perspective, perhaps the only articulation options that didn’t quite cut it for me with some of the instruments was a heavy staccato. The samples themselves have been

recorded very dry and without much by way of EQ or compression; those choices are left in your control. While you don’t have key switched articulations to deal with, you do have to roll your own sections; unlike many orchestral libraries, there are no ensemble patches here. If you want to build a brass section then you have to do it the hard way by combining a number of the solo instruments. However, that’s easy enough to do, and you can reduce the engine workload by unloading the samples from some of the microphones. And I have to say that, at this price, I think the sounds make the additional work worth the effort. The individual instruments sound pretty effective — there is a rather nice warmth to the tones — when played solo but are actually quite impressive combined into a section. The range of mutes all add the expected character so, whether you are looking to create orchestral‑style brass arrangements, dip into some soulful jazz, or even create some classic cartoon sounds, Model July 2016 / w w w . s o u n d o n s o u n d . c o m

Brass might well be worth a blow. John Walden $146.41 www.xtant‑

Wide Blue Sound Eclipse Kontakt Instrument Eclipse is a Kontakt instrument with a lot of similarities to Orbit, Wide Blue Sound’s previous opus. The installation process is an improvement on the early version I reviewed and now it resides snugly in Kontakt’s Libraries menu. It’s a little smaller too, taking less than 600MB of disk space. Intended as a source of atmospherics in motion, each Eclipse patch is built from four well‑crafted stereo samples that loop continuously; these are known as Orbits. The synthesis engine also includes filtering and transposition of each sample, and since the samples are rich and complex across the whole audio spectrum, filtering is a vital tool for homing in on sections of the samples ready to recombine with others. It means that, even though you’re given just 69 audio sources to play with, they can be


blended to form a large number of genuinely different permutations. The factory presets (stored as snapshots) provide excellent examples. The source waves are organised into three categories: Simple, Complex and Asylum. While there are some style overlaps, the Simple waves are primarily drones or Wavestation‑like shifting textures. They include many brooding, reverberant pads, sometimes with an added resonant whistle or bloated sawtooth bed. The Complex category is notable for its increased movement, dynamics and processing. Here you’ll find disturbing wails and reverse effects, plus the sounds of desolation, sensory deprivation (think ‘Altered States’)



and post‑apocalyptic gloom. Finally, Asylum hits the extremes like a blood‑splattered horror novel. This is the most diverse category and it includes computer‑like bleeps, deranged guitar loops, mechanical effects and tormented choirs. You can randomly scramble the available sources by clicking on the dice icon, but the real fun starts when you apply one of the three LFO shapes that govern the transitions from one orbit to the next. The shapes available are: pulse (a sawtooth wave), chop (a square wave) and flow (sine). Naturally you can set the rate of transition and the depth of the effect, and there are basic envelope controls too. It’s synthesis made simple, but I don’t mean that as a criticism. It’s surprising how many textures are ready to be discovered, and that’s even before you turn to the four‑part step sequencer and impressive effects engine, complete with convolution reverb. For extra performance kicks, Kontakt keys are provided to switch the sequencer on and off, mute individual Orbits July 2016 / w w w . s o u n d o n s o u n d . c o m

and so on. Eclipse is a high quality bundle of ready‑made textures for the film or TV composer. While the previous collection delivered cinematic ambience, Eclipse is notably darker thanks to its choice of source material. Yet there’s enough direct tweakability to serve up surprise twists long after you’ve become familiar with every source sample. It’s also cheaper than its predecessor (Orbit owners are entitled to a discount) and both are compatible with the forthcoming Lemur profile for iOS, known as Skypad. Paul Nagle $149

Big Fish Audio Country Guitars Multi‑format Country music can come in a rough and ready hoedown format, but that’s not what Big Fish Audio’s Country Guitars is about. The material in this particular sample library is radio‑friendly and absolutely ready for your next modern country rock chart assault.


Country Guitars provides around 3.5GB of 24‑bit audio samples/loops split across 20 construction kits and an ‘extras’ folder of additional loops. And while the guitar most certainly dominates, you get acoustic, electric, slide, baritone, banjo, ganjo (a six string banjo tuned like a guitar), steel and the occasional mandolin and resonator varieties thrown in for good measure. I explored the WAV loops but, depending upon which format you go for, you also get Apple Loops, REX or a Kontakt version. The package also includes pre‑configured DAW sessions for Logic, Pro Tools and Cubase, and these make it a doddle to get up and running with one of the construction kits in an instant. The kits themselves are extremely well stocked. You get a number of different song sections within each kit and multiple guitar parts for each song section. In terms of building your own arrangement, and adding variety as you progress through a full song structure, you are given plenty of options.


Country Guitars Whatever your personal take on country music as a genre, it is a musical world that is populated by some top‑notch musicians, songwriters and producers. That comes across in this library. The guitars themselves — whether acoustic or electric — sound absolutely fabulous. If you wanted a little tutorial on how to get that modern country rock sound (think Brad Paisley, Keith Urban or Dierks Bentley and you will be in the right ball‑park), this wouldn’t be a bad place to start. The electrics rock, but without going metal, and have bite, but without going twang. The acoustics have body but without getting July 2016 / w w w . s o u n d o n s o u n d . c o m

boomy. This is a bit of a master class in country guitar tones. The bulk of the kits are more mid‑ to up‑tempo, good‑time country rock, although there is the occasional dip into something that’s a bit mellower. OK, as with any construction kit sample library, you have to work within the confines of the musical structures you are provided but, unlike some such libraries (and some commercial albums), Country Guitars sounds like ‘all killer, no filler’ to me. Add some equally slick drums and bass — oh, and that top‑notch vocal talent — and these kits could slot right into any top‑10 country radio playlist. Media composers should perhaps note that the library comes within BFA’s usual restriction in terms of production music use (you need to apply for an additional license, although this is usually granted) but, whether for songwriting or media music use, Country Guitars absolutely nails the modern country rock guitar style and sound. John Walden $129.95

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scene for more than 50 years — and where there is music, there are recording studios. One Liverpool studio has been catering both to the local scene and visitors from abroad for three decades, and has been justifiably called “the most iconic studio in the North”. Following almost 30 years of constant operation at their current premises, and a history that goes back even further, this venture has played a defining role in the careers of some of the most influential British groups of the past decades. The Motor

Museum combines an excellent technical setup, a dedicated staff and a truly unique workspace which, probably, looks like no other recording studio in the world.

Larking About Originally conceived as a public transport repair depot, the building became home to the Lark Lane Motor Museum in the late ’70s. The collection of vintage cars and motorcycles on display apparently included the first Ferrari that was ever imported to the UK. It is located in

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the Lark Lane district in the South of Liverpool, which now carries a bohemian reputation as a nightlife hotspot. At the time, however, as studio founder Hambi Haralambous recalls, the museum was located too far away from the city centre to get enough visitors, and eventually had to close. Haralambous was a member of acts such as Tontrix and Hambi & The Dance in the early ’80s, and also part of a vital scene which included bands like Dead Or Alive and Frankie Goes To Hollywood. His first recording operation, The Pink Studio, started out in the basement of a Victorian villa in another part of Liverpool. The space was first used as Tontrix’s

“Motor Museum has played a vital role in the careers of several of the most influential UK bands of the past decades.” private rehearsal room, but after the band split up. it evolved into one of the first professional recording facilities in the city. After a brief stint in Italy, Haralambous was forced to leave the old premises in 1987, but the members of Echo & The Bunnymen — all motorcycle fanatics — pointed him towards the closing Lark Lane Motor Museum. After a year of

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planning and construcion, the studio reopened as The Pink Museum in 1988. The actual studio space is formed by a number of cabin-like rooms arranged within the vast 240‑square‑metre warehouse area. Construction was overseen by Phil Newell, who was also responsible for Virgin Manor and Townhouse: hence the latter’s notorious stone room, responsible for so many ’80s gated drum sounds, was replicated at The Pink Museum. The British reggae band Aswad were among the first customers at the studio, as well as members of Liverpool FC, who recorded the hit single ‘Anfield Rap’. In 1989 Andy McCluskey entered the studio for a three-day session which went so well that Orchestral Manoeuvres In The

Dark recorded their 1991 album Sugar Tax at the Pink Museum, spawning hit singles such as ‘Sailing On The Seven Seas’ and ‘Pandora’s Box’. Around 2000, when Haralambous decided to leave the music business in order to pursue a career as a film-maker, McCluskey bought the property and continued the studio’s operations. The torch was handed over to Mike Crossey in 2009, shortly after the original carpeted main live room was remodelled for a livelier, more ambient sound. Since 2012, the studio has been operated by producer Al Groves, who recalls: “One night we were out drinking in Liverpool, and my friend Mike Crossey told me that he was thinking of moving to London, and that the Motor Museum

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would be available. We spent a few weeks going over numbers before I sadly had to admit I didn’t think I could make the jump to take it over. Luckily, Mike had decided to stay for the foreseeable future, so the conversation ended. Six months later, I had a text from Mike telling me that he was due to leave at the end of the week, and had told Andy that I would be taking over! Immediately afterwards another text came from Andy McCluskey, and it was either take the big leap of faith, or regret it for a long time. I am really grateful to both Mike and Andy for giving me a little nudge to take it!“

Motoring On Today, the Motor Museum is centered around a Neve 8232 console with 32 channels, augmented by an outboard collection that includes pairs of Empirical Labs Distressors, ‘silverface’ UREI 1176LN and Cartec Program Equalisers. An Avalon VT‑737sp channel strip, a Retro Instruments Sta-Level and preamps by AMS Neve and API complete the picture. The microphone collection has been built around classics like the solid-state Neumanns U87 and TLM170 as well as numerous AKG C414s, yet it also includes

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more specialised mics such as a Manley Reference Valve Microphone, and ribbons like the AEA R44 and R88 Mk2, as well as a pair of Coles 4038s. One of the studio’s strengths is an instrument and backline selection that includes a wide choice of Fender, Marshall and Ampeg amplifiers along with keyboard instruments such as a Rhodes piano, Minimoog Model D and Octave Cat synths. Al Groves’ favourite, though, is a vintage 1960s Ludwig drum kit: “It always sounds great!” Motor Museum has played a vital role in the careers of several of the most influential UK bands of the past decades. Oasis recorded their first single

‘Supersonic’ here, and Arctic Monkeys used Motor Museum for some of their earliest recordings. Other seminal acts from this era include the La’s, the 1975 and the Coral. More recently Ben Howard, Jake Bugg, Tom Odell and Tribes have worked at the studio, and current clients include Bring Me The Horizon, Cast and Bellevue Days. These days, Al Groves is seeing a trend towards more specialised, producer-led studios. He concludes: “Being outside of London means I can have a world class studio but at a reallife cost, and a lot of the artists I work with love being able to come away to Liverpool to record. For them it’s like a working holiday!“

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Notes From The Deadline

TV Music From The Inside Sometimes a make-or-break opportunity can make you and break you...


ugust, six years ago. It’s a million degrees in my flat, as the sun streams in through the seventh-floor windows with lovely views across the Soho rooftops. There’s no air-conditioning to calm the raging heat blasting from the Mac Pro, two PC slaves, three big screens and 5.1 Barefoot monitors, and it feels like everything’s melting. I’m writing the first cue of my first movie. I’ve nailed the theme and have a cute, driving figure happening on the mandolin and ukulele. My clients often tell me they like the way my music drives the action along (particularly in factual TV, where there’s often not much action on screen), and at some point, this became a ‘thing’ that went from a happy accident to being part of my ‘style’. The thing about repetitive figures is they’re great, until they’re not. I’m working with a rough cut of the scene and it’s currently running at 4 minutes 35 seconds — about two minutes longer than it will end up. Four minutes of mandolin and ukulele would be enough to drive anyone mad, even if they weren’t playing the same little figure over and over again, and in the unbearable heat, even 30 seconds seems like too much. Nothing is working. I try modulating, but

it sounds cheesy. I try changing the pattern, but it feels interrupted. I try stopping it altogether, but it feels like the track has lost momentum. I simply don’t know what to do. I decide to Phone A Friend.

An Emotional Response Samuel Sim is probably in the middle of at least three projects that are far more important than my little indie film, but he listens patiently as I try to explain that I’ve completely forgotten how to compose, in fact, am not sure I ever knew how to compose, then start babbling as I bury my head in my hands in an all-too-familiar mix of shame, self-loathing and despair. He probably makes some very clever suggestions, perhaps involving the octatonic or maybe Schenkerian analysis, but I’m not really listening, because I already know there’s no possible way of writing myself out of this dead end, as I have no talent and the whole cue is rubbish. I go for a walk and a coffee. The temperature is much cooler at street level and I’m beginning to regain my composure, when the phone rings. It’s the producer of the big BBC drama series I spent the weekend pitching for. Actually, I finished my pitch on Friday, but during a family lunch on Sunday, I realised I needed to do

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an alternative end section, so I skipped pudding and went back into the studio. My head is spinning and I’m finding it hard to concentrate as he tells me he loves my demo and I’ve got the job. I’ve been desperate to make the jump from factual to drama, and for the last few years it seemed as if it would never happen. Suddenly I have my first movie and my first drama. By the time I get back to the studio the first episode is waiting for me and now I’m under pressure. As if that wasn’t enough, an old school friend sends over some tracks he’s been working on for a cookery series I’d asked him to help with, and they’re all bad.

Death’s Door I sort out the film cue in the next few hours (I leave the repetitive figure in the whole way through and it’s fine), then make a start on the drama. Around 2am I redo the cookery tracks, and

at 4am I open the sofa bed and grab three hours’ sleep. I repeat this every day (taking off Saturday afternoons and Sundays to spend time with my wife and two-month-old daughter) until some time in November, when, returning from the mix of episode four of the drama (most of which I sleep through, while Jake Jackson works his magic), I faint on the Underground and have to be helped off by a kindly young woman. I spend the next four days in bed, leaving just three days to write episode five.

This month’s guest column comes from film and TV composer Dru Masters, who does actually love his job. His credits include dramas Capital and Silk, Charlie Brooker’s A Touch Of Cloth, Panorama and, of course, The Apprentice. Dru has never won an award, nor even been nominated for one, but he’s not bitter.

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We examine the production of some recent hit records to help you brush up your listening skills. ZAYN ‘PILLOWTALK’ In past columns I’ve made disparaging use of the term ‘consonant sludge’ to describe a phenomenon that crops up periodically in bombastic chart-pop choruses, where so many harmonic instruments are layered on top of each other in pursuit of an epic ensemble sound that it’s all but impossible to hear any of them distinctly in the final mix — you just end up with a kind of amorphous chordal pad surrounding the drums and vocals. What I find quite refreshing about ‘Pillowtalk’, though, is how it turns this idea on its head, in a sense. What I mean is that there are plenty of chordal elements here that seem almost purposefully bland and pad-like (in particular the static power‑chord-style block harmonies of the chorus), but then the interest is maintained by swarms of fleeting sound-design elements that dart in and out of view from the cover of this

wash, frequently hovering on the boundaries of identification. Are those vocals bubbling up in the background before the first chorus (0:46-0:53)? Where do the vocal echoes of “climb on board” (0:15) end and the electronic blips begin? Is that an electric guitar or a synth at 2:40? So where a traditional consonant

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sludge takes lots of distinct instruments and swills them into an indeterminate mush, ‘Pillowtalk’ starts with something bland, and then seems to extract distinct instruments out of it, so to speak. Furthermore, while the resulting soundscape performs many of the same functions as the traditional sludge by maintaining an enveloping texture and hinting at both epic space and innumerable instruments, at the same time it draws out masses of sonic detail to entertain and intrigue the repeat listener in equal measure. Such production refinements lend proceedings a definite air of artistic maturity here, in my opinion, and can hardly have harmed Mr Malik’s campaign to draw a firm line under his boyband past. Mike Senior JONAS BLUE ‘FAST CAR’ What if you want the sound of vocal double-tracking in your mix, but the singer’s only recorded the line once? Well, one time-honoured approach is to use modulation treatments such as single-tap chorus (ie. a super-short delay line with a modulated delay time) or vibrato to create a clone of the recorded vocal with minute pitch/timing variations — akin to those you’d encounter in a real double-track. The challenge is differentiating the clone sufficiently to create a passable illusion of a second performer (rather than just a hideous plastic chorusing effect), but without making the timing/tuning of the whole confection seem sloppy. Another tactic that’s gained currency more recently is to process the vocal clone with repitching software such as Auto-Tune or Melodyne

to give it a different pitch-profile. Although this avoids compromising the overall tuning or timing, the danger is that you end up with rather a phasey sound when the clone combines with the original, because the two parts remain too similar. In practice, a combination of these two basic techniques (whether applied from first principles or via a dedicated ‘vocal doubler’ plug-in) is sensible, in my view, with the modulation treatments introducing variation, and the pitch-processing keeping the tuning fluctuations within the bounds of taste. Which brings me to the vocal sound in this recent Tracey Chapman cover. On first impressions it has the flavour of double-tracking, but the match between the two parts is so tight that I suspect it’s a clever emulation. It’s a tricky one to call, admittedly, because super-tight audio editing of real double-tracks might also have generated such precise synchronisation. On balance, however, a couple of bits of evidence swing

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the argument in favour of the artificial explanation for me. Firstly, compare the right and left channels of the mix on the word “car” at 2:09. While the right-hand layer sounds fairly natural, the left-hand one exhibits weird artifacts that I strongly associate with artificial pitch-modulation. And, secondly, consider the subtle octave doubling that’s lurking in the shadow of the main vocal line most of the time. (If you’re having trouble hearing it, check out “Momma went off and left him” at 1:59, where it’s particularly audible.) Such a synthetic sound must surely derive from pitch-shifting technology, and if pitch-shifting was used there, then I reckon it’s odds on that the double-track was generated in a similar manner. Mike Senior ALAN WALKER ‘FADED’ This nicely stripped-back example of the Euro-EDM mix-pumping cliché makes a clear distinction between the textural elements of the backing track (ie. the bass, synths, and sound effects), which all duck heavily in response to the kick-drum, and the vocals, which don’t. Clearly this wouldn’t be possible if the pumping effect were being created with a simple master-bus process, so if you’re trying to emulate this kind of sound then you need to find a way to implement the pumping on a track-by-track, or at least bus-by-bus, basis. Furthermore, we’re also treated to the hoary old trick of pumping triggered in the absence of the kick drum, for example on the bass line during build up to the first chorus (0:42-0:54) and on the whole backing mix at 1:15 and 2:52. Fortunately, there are now myriad ways

of implementing a sufficiently controllable pumping scheme in modern DAWs — my favourite being to use duckers on the appropriate channels, and then feed their side-chains from a dedicated trigger track (usually an edited copy of the kick-drum audio) which doesn’t itself feed the mix. What I would advise against, though, is using MIDI controllers and/or track automation curves for pumping purposes, because it’s trickier to make per-channel adjustments to the ducking characteristics that way, or indeed to quickly experiment with different instances of kickless modulation. Mike Senior KENDRICK LAMAR ‘UNTITLED 02: 06.23.2014’ There are so many ideas bursting out of Lamar’s recently released demo compilation, Untitled Unmastered, that it’s hard to know where to start when it comes to writing about it! For my part, the track ‘Untitled 02’ is probably the highlight, if only for the rapper’s

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tremendously inventive use of vocal deliveries and textures: the full-bodied “get God on the phone” at 0:39 versus the following verse’s pinched breathiness and quasi-yodelled phrase endings; the chorusey falling group interjections which begin at 1:14; the strangely robotic triplet-based riff at “pimping and posing”; the emphatic vibrato in the backing-vocal “ah” at 1:40; the laid-back patter of that extended third verse from 2:30; and the strung-out higher-register from “what if I certified” at 3:33. There are plenty of rappers who struggle to match that expressive range on a whole album, let alone one demo! I’m also a sucker for this mix’s method of generating an exaggerated depth illusion, where a massive distance is implied between the up‑front elements (the point-source lead rap, super-dry, short-release kick, and wide stereo percussion) and background textures which seem to be made up primarily of a piano’s recirculating reverb. Those good things notwithstanding, I can’t help feeling that this album’s title is missing a word: uncredited. There’s a limit to how seriously I can take biting social and political commentary from someone who releases an album full of collaborations with a slew of talented

fellow musicians, engineers, and producers, but who steals their credit by putting nobody’s name on the CD but his own. What’s worse is that if the sense of mystique in this specific instance actually drives greater recognition to these people than might otherwise have been the case, it’ll provide lazy/unscrupulous artists of the future a golden opportunity to weasel out of their own moral responsibilities by arguing that it’s actually better press to leave the credits off. Oh yeah? In which case, leave the bloody artist’s name off too! Mike Senior

Classic Mix DAVID BOWIE ‘STARMAN’ (1972) As befits a song with the line “hazy cosmic jive” in it, this production is replete with sonic oddities. To be fair, the super-damped, close-miked drums

and wafer-thin cymbals were fairly commonplace in productions of the era, but I’ve always found such wide tom panning a strange listen, especially from audience perspective, as it is here, with the low tom hard left — after all, the only person who normally appreciates

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that kind of panning width on drums is the drummer, not the audience. And take another listen to the acoustic guitar in the right-hand channel during the introduction. That’s a load of pick noise in there! But of course it’s an indication of the production acumen at work here that such individualistic sounds slot together in the way they do. So, for example, with the cymbals so low in the balance, the pick noise from the two guitars ends up filling that arrangement role in a manner akin to stereo percussion; and the claustrophobic closeness of the miking on both the drums and the acoustic guitars maximises the arrangement contrast when the more expansive string and electric-guitar sounds enter later in the song. In addition, though, there are a couple of fun little technical details to be discovered. The first is just before the first drum fill at 0:19, where the lefthand side of the stereo image seems to dry up about a second before the first drum hit. I wonder whether, in the absence of a right-channel guitar double-track during the introduction, the mix engineer (the legendary Ken Scott in this case) decided to even up the stereo balance by manually fading up the lefthand channel of the plate reverb before the start of the track, and what we’re hearing is him fading it back down to a more appropriate mix level a little too early before the drum fill. Whether by luck or design, though, I rather like the way this creates a sense of momentary

vacuum in the texture, as if the production is holding its breath briefly in anticipation of the drums’ arrival. The second thing that piqued my interest was that Ken Scott has mentioned in interview that he only remembers mixing one version of this song, but there are two different versions available: an album version and a single version, the latter featuring a much higher level of the phased piano/ guitar pre-chorus fills at 0:51 and 2:10. To investigate, I loaded the 2012 remasters of both versions into my DAW, matching both the playback speed and polarity (inexplicably inverted in the single version) and, judging by the nature of the phasing artifacts between them, I suspect that both are in fact the same mix, except that the single version has had both those aforementioned two-bar sections spliced into it from a different mix. Indeed, if you examine the waveforms at the starts of both those choruses, it appears that an orphaned fragment of extra transient has been spliced in from the alternate mix just before the downbeat, and in the second case you can clearly hear its little ‘pop’ in the left-hand channel (at 2:15). Moreover, while I had the editing scissors out, I discovered that the first and second two-bar fills phase against each other if you line them up, whether you try it with the album or single versions — in other words, it appears that the second fill is a copy of the first in both mixes. Mike Senior

July 2016 / w w w . s o u n d o n s o u n d . c o m


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What’s behind Conor Maynard’s vocal sound?

I was listening to Conor Maynard’s YouTube cover of Shawn Mendes’ ‘Stitches’ ( watch?v=Jnm4UMEQFyc), and wondered if you could provide any insight into what he used to get that huge, rich vocal tone. Any time I record vocals over a piano backing I seem to end up with a harshly bright or ‘present’ vocal that does not blend with the piano at all. I use the Synthogy Concert D piano software instrument, which I love the tone of, but just cannot seem to blend it nearly as well as this example. I am sure there

is compression, reverb, and possibly delay being used but the amounts or specific tricky settings are what I am really curious about. SOS Forum post Mike Senior replies: Well, the first thing to point out is that we’re not hearing Maynard through the microphone shown in the video itself — his line has clearly been recorded separately. (If you need any convincing, check out 2:19, where the audio track and his mouth disagree on the first word of “I’ll be needing stitches”.) That said, the large-diaphragm condenser mic shown in the video might well have been used for the recording session, as

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the tone is very much what I’d expect from a microphone like that. If you listen to the word “deeper” at around 0:31, the ‘p’ produces a bit of a ‘pop’, so I’d guess that Conor was working close enough to the microphone that the popshield was unable to sufficiently block the wind-blasting of his plosives — in other words, a distance of eight inches or fewer. This surmise is further supported by the richness of the vocal’s low end, which would have been strongly boosted by the proximity effect at that close range, and the audibility of extraneous mouth noises (eg. on “(be)fore” at 0:19 or “sore” at 0:26), which don’t project nearly as well as other elements on a sung performance. I suspect that the mic was placed directly in line with his mouth too, because plosive blasts tend to be pretty directional, and we probably wouldn’t have heard them had the mic been more off-axis. This kind of mic position is something of a mixed blessing in terms of a vocal’s high spectrum, however: on the one hand it typically captures plenty of intimate, pop‑tastic breathiness; but on the other hand it tends to overemphasise the sibilants (‘s’ and ‘sh’ sounds), fricatives (‘f’ and ‘th’ sounds), and noisy stop-consonants (‘t’ and ‘k’ sounds) as well. So one of the primary concerns when recording vocals of this type is how you deal with that. One approach is to get the singer to replace these sounds with less harsh-sounding versions — for example, adjusting ‘s’, ‘f’, and ‘t’ sounds to more closely resemble ‘z’, ‘v’, and ‘d’. Although you might think it would quickly make nonsense of the lyrics, in practice listeners

Q&A are remarkably tolerant of this in practice. The exact choice of mic can make a big difference too, as most large-diaphragm condensers boost the high end of on-axis sounds, and the exact nature of this boost can vary a great deal between models. Like almost every other mainstream pop vocal these days, this has almost certainly been buffed with judicious pitch-correction processing. If you don’t normally pitch-correct your vocal recordings, then this may be part of the reason they don’t seem to blend as well as this — pitching isn’t just a musical issue, because sounds that are in tune also sit better in the mix. It’s also clear that the level of the vocal has been extremely tightly controlled at mixdown, as likely as not by a combination of fairly heavy compression and detailed fader automation, and again this level solidity contributes to the sense that the vocal blends naturally with the piano — for example, it never seems to leap out further in front by being momentarily too loud. As you’ve probably already realised, this kind of vocal sound needs additional high-frequency emphasis at mixdown, but the difficulty with that is avoiding overemphasis of the noisy consonants (especially sibilants) at the same time, so another key task is careful consonant management. If the recording’s been carefully done, a de-esser may be all that’s required to rebalance things, but in practice many mix engineers employ much more labour-intensive methods where maximum vocal brightness is required. For example, some will edit all the consonants onto a different track, and then boost the high frequencies only the

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consonant‑free track, while others use fader and EQ automation to adjust the character of each consonant manually. If you give all those steps enough time (and they do often take a while!), I don’t think you’ll actually find too much difficulty getting the vocal to blend in the final mix. Just a touch of feedback delay and plate reverb (as in Maynard’s case) should give you all the ‘glue’ and sustain you need. Do consider de-essing the sends to both of those effects, though, and EQ the reverb return to avoid a build-up of muddy low-mid range, given that the dry vocal has a lot of energy in that region.


Can PSU voltage be ‘stepped up’?

I’ve read so many people saying that a 9V power supply for guitar pedals means they can’t give you enough headroom, and the same about the 500-series 32V (±16V) supply. But surely there’s a way designers can ‘step up’ the voltage inside a device. After all, I’ve seen units running on 24V DC supplies or off USB ports or even 9V batteries that can give me 48V phantom power. So why can’t they all use a standard power supply and do that, to give higher voltages inside if they need them? Are significant trade-offs involved — and is there any reason to avoid units that do this? Douglas Bannister via email SOS Technical Editor Hugh Robjohns replies: The short answer is yes, voltage can be stepped up, and there are a variety of techniques available to achieve that depending on whether we’re talking AC or DC, the amount

Q&A of increase needed, and the power consumption requirements. Today, a lot of equipment employs ‘DC-DC converter’ chips, which can synthesise almost any required DC voltage from any other with remarkable efficiency, but inevitably such devices add to the equipment build cost, consume power, and generate heat, all of which limit their practicality in some applications. Although relatively rare today, they can also cause interference in some cases, which isn’t what you want in a high-quality audio device! While it could be argued that the ideal approach is always to generate the required power‑rail voltages directly from a mains supply, that’s obviously not always possible. Modern DC-DC converters offer a mature and effective technology, which makes operation from batteries or low-voltage supplies very practical in many circumstances, and I see no reason at all to avoid such devices. The headroom issue warrants a little closer investigation. The power-rail voltage for audio circuitry inherently sets the limit for the maximum signal voltage that can be processed; any signals that try to go higher will clearly become ‘clipped’. As audio signals swing above and below a nominal mid-line reference level, a 9V supply allows for a maximum voltage swing of ±4.5V, but because of the way active components work in audio circuitry we can’t usually access that whole range. For the sake of argument, let’s assume a 9V-powered device supports an audio signal with a swing either side of the mid-line of up to ±4V. Converting that into a more meaningful signal level in terms of dBu values is slightly complicated because it depends on the

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Q signal’s ‘crest factor’ (the ratio of the peak value to the RMS value). If we’re talking sine-wave tones, then the crest factor is 3dB, but for typical music the peaks can easily be 12-20dB higher than the RMS value, and that’s why we need headroom in our audio equipment! (See diagram.) So, a sinewave signal with a 4V RMS value equates to a signal level of just over +14dBu; the calculation is =20*log(4/0.775). However, the peak level of a sinewave signal is 3dB higher than its RMS value, and the peak level matters here because it’s going to be restricted by the power rails. So a ±4V working range supports a sine-wave signal amplitude of up to about +11dBu (14 minus 3dB), and for a really peaky signal (such as from a guitar!) the maximum supported signal level before clipping could be as low as -6dBu (14 minus 20dB). The output from an electric guitar is


generally classified as ‘instrument level’ which is specified as a nominal -20dBu, in which case we would appear to have around 14dB of headroom (20 minus 6) which isn’t bad… but it’s not great, either, and with more efficient pickups and more percussive playing styles we could easily find ourselves with surprisingly large transient signals that could well start to clip. For this reason, some manufacturers prefer to employ higher power rails in their equipment to afford greater headroom margins. Professional equipment generally supports a maximum interface level of +24dBu (measured with sine-wave tones), but again taking the crest factor into account, the peak voltage of a +24dBu tone would be just under 17.5V. From that, we can see that the equipment power supply must be at least ±18V, and in practice it’s often ±22V or even higher.

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You mentioned the ±16V rails of the 500-series racks, so let’s consider the maximum signal levels that can be supported there. If we assume the maximum voltage swing in a device powered in a 500-series rack is ±15.5V, the highest sine-wave signal level that could pass without clipping would be around +23dBu, which is surely good enough for most situations. In the few cases where a higher output signal level was needed, one simple solution would be to use a step-up output transformer, providing some ‘free’ signal voltage gain! So while having ±16V power rails is inherently slightly restrictive compared with ±22V rails, say, the practical impact probably isn’t that significant in most applications, can careful and sympathetic gain structuring will easily avoid any clipping issues anyway. One final observation along those lines: the Neve 1073 preamp/EQ module operates on a single +24V supply rail, allowing a maximum internal signal level of around +20dBu — but with transformer outputs! If it was good enough for Neve...


Do I need a shockmount?

I would just like to ask if it’s important to use microphone shockmounts? SOS Forum post Hugh Robjohns replies: I think this is one of those, ‘it depends’ answers. If the mic is not subjected to external mechanical vibration then shockmounting is unnecessary, clearly. And in many project-studio situations there won’t be any significant mechanical vibrations to worry about. Moreover, if what you’re

Q&A recording has no relevant low-frequency content you can use a steep (18dB per octave) high-pass filter to remove any unwanted LF rumbles anyway. So again, no shockmount is necessary in that kind of situation. However, as a rule of thumb, the larger the diaphragm the more susceptible the mic will tend to be to mechanical rumbles (think long ribbons and large-diaphragm capacitor mics). Also, the more directional the polar pattern the more susceptible the mic will be to LF nasties, too, so figure-of-eight, hypercardioid, and cardioid, in that order. Subcardioids and omnis are far less prone to picking up unwanted mechanical vibrations. The reason for these trends is that large diaphragms generally, and the diaphragms of directional mics, tend to operate with relatively low tensions, making them far more prone to wobbling about in response to external vibration. It’s often recommended to always use a shockmount, and I do generally do use shockmounts everywhere, mainly because I have good ones to hand, so it would be a waste not to! But a poor or inappropriate shockmount can make the situation worse rather than better, so some caution is required. It may seem counterintuitive, but you may well get better results with no shockmount than when using a poor one! A shockmount works as a mass-spring damping system, and is not too dissimilar to the suspension on a car. All systems of this kind have a resonant frequency, and above that frequency (about three times higher, in fact) the damping system will act to remove vibration very effectively. Below the resonance, though, it won’t,

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and at the resonance itself the system will actually tend to amplify the vibration! Consequently, to be genuinely effective, a shockmount system must be designed to have a resonant frequency somewhere below 6-8 Hz, so that it able to provide effective damping by the time useful audio information starts to appear around 20 or 30 Hz. Lots of factors affect the resonant frequency of a mass-spring system, but for a given spring compliance the obvious variable in the context of a microphone shockmount is the mass of the microphone it’s being asked to support. Consequently a shockmount must be designed and specified to work with a specific microphone, or at least a narrow range of mic weights. Sadly, the cheaper end of the mic market seems content to supply simple generic shockmounts, with no regard to optimising the complete system’s resonant frequency. The results are, therefore, somewhat unpredictable. This is especially true of the ‘cat’s-cradle’ style of shockmount for large-diaphragm capacitor mics, the weights of which

Q&A can vary enormously. I am a big fan of the Rycote InVision shockmounts which I use in various models almost exclusively for all my small- and large-diaphragm mics. I’ve found these to be superbly effective, fundamentally because they’re designed properly; the compliance of the suspension Lyres in each model is adjusted through the manufacturing process to suit the weight of the specific microphone being supported, and the company website specifies different InVision models for different sizes and weights of mics. Inevitably, the InVision shockmounts are not pocket-money cheap, but they will last a lifetime and you’ll never have to replace a snapped elastic or rubber O-ring! If you don’t have access to appropriate shockmounts, it’s always worth experimenting with a layer or two of carpet, or some furniture foam cubes, placed under the feet of your mic stand’s tripod legs. This should help to curtail the worst mechanical thumping of footsteps and the like from reaching the mic. It’s always instructive to use a spectrum analyser (often provided as part of multi-band EQ plug-ins) to examine what’s going on below 40Hz on the mic signals, too. Ideally, there shouldn’t be anything there... so if you can see a lot of energy it will be worthwhile fitting a decent shockmount and/or applying a steep high-pass filter. (Bear in mind, too, that extreme LF signals can be generated by things other than direct mechanical vibration, such as drafts and air currents around the mics, and acoustic LF wavefronts from nearby machinery or traffic).

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SOUNDING OFF We need a royalty revolution! PETE BRAZIER


fter 28 years making a living by writing, producing and releasing music, getting paid what I’m due gets harder every day. Streaming services have changed the landscape a bit, but they’re not the main culprits. (Sure, they might fail to pay a reasonable share to the artist/label, they list unreasonable terms, and we have to take on trust the number of streams they declare, but we can choose not to use them.) No. The chief problem is that the process for royalty collection is laborious, complex and at times hostile to the artist. Unfortunately, this is how the ‘industry’ likes it: why would they want to spend the £70 million they take in administration costs chasing the hard-earned royalties of small independent artists when they’re under almost no obligation to do so? Take this recent experience, which should have been oh‑so simple. My co-writer and I owned a song and its recording

that, with a new vocal and some mix touches, found its way onto an American release which sold around 800,000 units; I was chuffed. Puzzlingly, though, no royalties were forthcoming. It transpired that a lot of the record sale receipts had been eaten up by unexpected ‘recoupable costs’, including £80,000 per month to maintain a web site that remained dormant. If that wasn’t bad enough, the balance of royalties wasn’t paid by the American label to the UK collection society — because there’s no mechanism to ensure missing royalties are chased up — and the band with the most-played radio record in the US were left completely broke. You might be forgiven for thinking the label were committing fraud. (I couldn’t possibly comment). But the most frustrating thing is that the collection society were under no obligation to lift a finger to help me. I was left in an unpaid stalemate. To make matters worse, I was collecting my co-writer’s share on behalf of his widow and July 2016 / w w w . s o u n d o n s o u n d . c o m

About The Author

Pete Brazier has over 25 years’ experience in the music industry as an artist, writer and producer, working for many large labels including Warner Chappell, Zomba, Parlophone, Arista and a short stint signed to Simon Cowell. He has run the Vertical Rooms label and studio for 15 years and also lectures on the music industry and music production. the collection society sent her a form to collect his royalties posthumously; the form was titled ‘Post — Hummus’. My late co-writer’s preferences relating to chick-pea-based dips are not my affair, but if the society’s paperwork never sees a spellcheck or a proof reader, how diligent do you think they are in chasing your money? When, eventually, my

SOUNDING OFF live-performance payments collections process, where were calculated, I decided an artist can input their to answer that question. It song name, lyrics, co-writer turned out that the figures details, ISRC, UPC, WAV were based on information and MP3 attachments and gleaned during a brief so on, and submit the lot telephone conversation directly to the collection with the band’s lead singer. society and any other No need to register your places with a single mouse songs, then — just make click? Such a system would, sure bands keep a tally of course, save artists of how many times they and businesses countless play them! Staggered by hours and money; that the unprofessionalism and a universal system like ineptitude, I was left with this doesn’t exist in 2016 no option but to accept beggars belief. (YouTube the royalties and shake my have recently promised head in disbelief. something along these More recently, at lines, which is a step in the a collection-society event, right direction, but it’s only I witnessed a young for YouTube!). The ‘expert’ The return the Minimoog. musician posingof the panel passed the question Moog Music are about to remake history with theamongst reissue excellently-dressed panel themselves of the Minimoog Model D, but how will it measure up to a good, clear question. ineffectually before its illustrious forebear? Find out in our extensive review in the next issue of Sound could On Sound. Where, he asked, he suggesting that our young find the software system friend keep a spreadsheet on sale Thursday 21st July. toAugust assistissue artists with the of all his practices and Available at WH Smith and all good newsagents. registration, upload live jams! Like the worst Never miss an issue: subscribeand at


NEXT MONTH IN The return of the Minimoog. Moog Music are about to remake history with the reissue of the Minimoog Model D, but how will it measure up to its illustrious forebear? Find out in our extensive review in the next issue of Sound On Sound.

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politicians, they dodged a simple but uncomfortable question and answered a different one entirely. This all needs to change. The process can be simplified and made accessible to artists like you and me, but collection societies are never going to develop the software to make this happen. We writers and performers need to stand together and make these money-hungry companies accountable for what they are doing with our hard-earned cash. Who’s with me? If you would like to air your views in this column, please send your submissions to soundingoff@ or to the postal address listed in the front of the magazine.

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