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SOUND ON SOUND THE WORLD’S BEST RECORDING TECHNOLOGY MAGAZINE

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1985 — 2020 TM

MUSIC PRODUCTION TECHNIQUES / INDEPENDENT IN-DEPTH PRODUCT TESTS / ENGINEER & PRODUCER INTERVIEWS / LIVE SOUND

HOW TO

CHOOSE AN

SEPTEMBER 2020

AUDIO

INTERFACE www.soundonsound.com

Studio One 5 PreSonus tempt the pros with massive update

Make your studio artist-friendly Control modular synths from your DAW! VOLUME 35 • ISSUE 11

SPITFIRE AUDIO INSTRUMENTS WORTH £1697

www.soundonsound.com

REVIEWS: APPLE / ARTURIA / LIQUIDSONICS / CRANBORNE AUDIO / RUPERT NEVE DESIGNS / IK / OUTPUT / AKG / DDMF / NOVATION / VSL

TECHNIQUE: MANAGING SIBILANCE / REMOTE REHEARSAL WITH JAMULUS / DAW WORKSHOPS

September 2020 £5.99

LEADER

CHOICES CHOICES W

hy is Sound On Sound publishing a major article about choosing an audio interface? Many reasons. One is that audio interfaces are now the most universal item of studio equipment there is. It’s possible to make music without a microphone, or without loudspeakers, but who now does so without a computer? Another is that choosing an audio interface is difficult. Simply identifying your own present and future needs is hard enough; deciding which product best meets them is even harder. There are hundreds, possibly thousands of devices on the market, and it’s not always obvious what separates them. Yet another reason is that buying an audio interface is not like buying a guitar or a synthesizer. It’s not only a difficult decision, but a boring one. The device we use to get audio into and out of our computer isn’t an inspirational one that can help us reach new heights of creativity. Not many of us want to be comparing spec sheets or counting input connectors when we could be making music. This is where Sound On Sound can bridge the gap. It’s our job to know what musicians shouldn’t have to learn in order to make music. And it’s our mission to present that information in ways that make difficult choices easier. We won’t pretend there’s a simple answer to the question ‘What’s

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the best audio interface for me?’, because that would be silly. It’s a complex question, and like most complex questions, it’s best answered by breaking it down into smaller questions that are more easily answered, as we’ve done in this month’s cover feature. However, researching the feature also brought home to me that choice can end up feeling like an illusion. There are thousands of electric guitars on the market, but how many are not derived from a 60-year-old Fender or Gibson template? Likewise, I can buy a 1U rackmounting interface with eight mic preamps, a couple of headphone amps and an ADAT port for expansion from dozens of manufacturers. But what if I want more than two headphone outputs? What if the ideal number of preamps for me is 10, or 12? What if I own no other rackmount gear, and would prefer a desktop unit? What if I want transformer-balanced preamps that can be pushed for a bit of saturation and colour? What if I’d like an audio interface that can host 500-series modules, or one with an integrated patchbay, or DAW transport controls, or one that is optimised for driving modular synths? In an increasingly crowded market, perhaps there are still untapped opportunities for manufacturers to make their products stand out from the herd — and make choosing an audio interface less boring?

ADV ER TISIN G

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“In an increasingly crowded market, perhaps there are still untapped opportunities for manufacturers to make their products stand out from the herd.”

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The contents of this publication are subject to worldwide copyright protection and reproduction in whole or part, whether mechanical or electronic, is expressly forbidden without the prior written consent of the Publisher. Great care is taken to ensure accuracy in the preparation of this publication but neither Sound On Sound Limited nor the Editor can be held responsible for its contents. The views expressed are those of the contributors and not necessarily those of the Publisher or Editor. The Publisher accepts no responsibility for the return of unsolicited manuscripts, photographs, or artwork. © Copyright 2020 Sound On Sound Limited. Incorporating Music Software magazine, Recording Musician magazine, Sound On Stage magazine, SPL magazine, Sound Pro magazine and Performing Musician magazine. All rights reserved. All prices include VAT unless otherwise stated. SOS recognises all trademarks.

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124 INSIDE TRACK

IN THIS ISSUE

www.soundonsound.com

September 2020 / issue 11 / volume 35

FEATURES 30 Kaitlyn Aurelia Smith We talk to prolific producer Kaitlyn Aurelia Smith about her ambient compositions and love of Buchla synths.

48 Setting Up & Using Jamulus

WIN

SPITFIRE AUDIO INSTRUMENTS

WORTH £1697

The free and open-source Jamulus app lets you play along with other musicians over the Internet in real time.

68 The Artist-friendly Studio Ten tips to keep musicians ‘in the zone’.

82 Pigs Pigs Pigs Pigs Pigs Pigs Pigs:

Recording Viscerals

The rules of heavy metal are well established, but for their latest album, Pigsx7 guitarist and producer Sam Grant added a few of his own.

90 Managing Sibilance

PAGE 26

Prevent pesky ‘ess’ sounds from ruining your tracks.

96 How To Choose An Audio Interface The audio interface is at the centre of every modern studio, but how do you choose the right one for your studio?

124 Inside Track: Ariel Rechtshaid

& Rostam Batmanglij

104 Digital Mixers In The Studio

Haim’s latest album was made with a mix of old-school and modern techniques and equipment.

Many of the features that won live sound engineers over to digital consoles also have huge potential in the studio. We round up some of the best.

158 Q&A

112 Modular Interfacing

162 Why I Love... The JBL Control One

Everything you need to know to get your DAW talking to your modular system via your audio interface.

Sound designer Helen Skiera explains her love for a small but plucky loudspeaker.

Your studio and recording questions answered.

96 HOW TO CHOOSE AN AUDIO INTERFACE

ON TEST 8

IK Multimedia iRig Keys 2 Mini

74

Controller Keyboard

10

AKG Lyra Multi-pattern USB Microphone

12

Yamaha THR10 II Guitar Amplifier

14

Liquidsonics Cinematic Rooms Reverb Plug-in

18

Cranborne Audio EC2 Dual-channel Mic Preamp

22

DDMF Magic Death Eye Compressor Plug-ins

38

PreSonus Quantum 2626 Thunderbolt 3 Audio Interface

42 52

VSL Big Bang Orchestra

Output Thermal Parallel Distortion Processor

94 Plugin Alliance Brainworx

Amek EQ200

EQ Plug-in

116 Arturia AudioFuse Studio USB Audio Interface

120 Novation Launchkey MkIII Controller Keyboard

130 Apple Logic 10.5 DAW Software

PreSonus Studio One 5

Rupert Neve Designs RMP-D8 Eight-channel Dante Mic Preamp

142 Neunaber Wet Reverberator Reverb Plug-in

144 Sugar Bytes Looperator for iPad Step-based Multi-effects Processor

146 Sample Libraries Sample Logic Cinematic Guitars Motion Orchestral Tools Tableau Solo Strings Sonic Atoms Baltic Shimmers Ergo Kukke Trails

140 Apogee Clearmountain’s

Spaces

Scale Detection & Generation Plug-in

64

SubBass Doctor

Reverb Plug-in

Plugin Boutique Scaler 2

142 Soundevice Digital Sub-bass Generator Plug-in

Sample Libraries DAW Software

60

88

Yamaha CP88 & YC61 Stage Keyboards

140 Eventide Undulator Modulated Tremolo plug-in

141 IK Multimedia Z-Tone Buffer

Boost & Z-Tone DI

Variable-impedance Guitar Preamplifiers

WORKSHOPS 148 Digital Performer 152 Reason 154 Pro Tools 156 Cubase

ON TEST

PAUL WHITE

E

ssentially a cut-down version of the three-octave iRig Keys 2, the iRig Keys 2 Mini is, as it suggests, a mini-key-format MIDI controller keyboard designed for mobile use. It features 25 velocity-sensitive keys, but loses the pitch and mod wheels and the external pedal input of its larger siblings. In addition to offering laptop/desktop computer USB compatibility, the iRig Keys 2 Mini can connect directly to iPhones, iPads and many Android models. Its data connection is via a mini-USB port on the rear panel and the included adaptor cables convert this to USB-A, Lightning or Android connectors. Power comes from the connected device, but it may also be powered from a USB power supply or USB battery pack. There’s a TRS 3.5mm jack audio output to connect speakers or headphones, which is particularly handy if you have one of the newer Apple devices where the headphone output jack has been omitted ‘for your convenience’. There’s one five-pin MIDI In/Out adaptor cable included that connects to either the mini-jack MIDI In or MIDI Out rear panel sockets, enabling use as a standalone MIDI controller or MIDI interface. Additional cables may be purchased separately. As with the other

8

IK Multimedia iRig Keys 2 Mini Controller Keyboard

Need a flexible, on-the-go keyboard? Look no further... models in the series, there are additional panel controls — including volume control, a data push/turn encoder, four assignable knobs, program up/down and octave switching. Edit mode, entered by pressing both Octave buttons, is where you assign the controllers and other settings such as MIDI channel and so on. When editing, the keyboard keys are also brought into use with their functions printed above them. Other than the four Set LEDs and the backlit buttons, there’s no display. Edit changes can be saved as up to eight Sets for later recall. A selection of music-making apps is bundled with the keyboard, comprising SampleTank 4 SE with upwards of 2000 sounds, SampleTank Free for iPhone and iPad, iGrand Piano Free for iPhone, iPad and Android, and iLectric Piano Free for Android.

September 2020 / w w w . s o u n d o n s o u n d . c o m

While the three octave iRig Keys 2 would be my preferred choice purely because it retains the mod/bend controls and footswitch input, I can see the appeal of the iRig Keys 2 Mini for use on holiday or while travelling. It’s small enough to fit into a backpack or briefcase, it works with mobile devices as well as computers and it covers all the essentials. IK’s bundled software sweetens the deal further, so if compact and functional is what you need, the iRig Keys 2 Mini fits the bill nicely.

summary A practical and compact solution to portable music-making, so long as the lack of those facilities listed above is not a deal-breaker.

££ £105.99 including VAT. WW www.ikmultimedia.com

SM6 The SM6 Range Wh atever your genre, styl e o r s pace D el iver a mix worthy o f your mu s ic.

Distributed in the UK & Ireland by SCV Distribution 03301 222500 | www.scvdistribution.co.uk

ON TEST

AKG Lyra

Multi-pattern USB Microphone This classy USB mic offers a useful range of polar patterns.

PAUL WHITE

U

SB microphones are nothing new but there’s a lot more to the AKG Lyra than its gloriously vintage look, which tips a nod towards both vintage broadcast mics and AKG’s own C414. While clearly of appeal to podcasters and videocasters, the microphone also has applications in home or mobile music recording. The Lyra sits atop a custom all-metal desk stand, and the swivel section can be separated by using a coin to unscrew the threaded thumbscrew that secures it to the base of the stand. The mic and swivel section may then be mounted on a standard mic stand, and an EU thread adaptor is included. Built into the case is a shockmount for the capsules and a sound diffusor to help reduce plosive pops. Power comes from the USB port into which the mic is plugged; a suitable USB cable is also included. For those who haven’t yet decided on which DAW to go with, a copy of Ableton’s Live 10 Lite is included. Perhaps the most noticeable departure from convention is that the Lyra actually incorporates four back-electret capsules, two angled slightly away from each other and forward facing, plus two rear-facing capsules angled the same way. AKG call this their Adaptive Capsule Array. A four‑way rotary switch then selects between four pickup patterns depending on the application. Position one is for a single user sitting in front of the mic and yields a cardioid polar pattern, while position two accommodates a pair of performers, one behind the mic and one in front. This appears to be an omni pattern. Position three picks up from the front of the mic in a tight stereo pattern, while position four picks up both front and rear in a wide stereo pattern for occasions when more ambience is needed. A headphone amplifier with zero‑latency source monitoring is built into the microphone, the headphone connection point being a 3.5mm stereo

10

jack at the base of the unit next to the USB connector. Don’t forget to mute the source audio monitoring in your DAW though, or you’ll hear both the latency-free signal and the slightly delayed one that has done the round trip to and from your DAW. There’s a rotary headphone level control on the front of the mic, along with a mute button that kills the mic without cutting off the DAW playback. Four LEDs show which mic pattern is selected. On the rear panel, alongside the pattern selection knob, there’s a control for microphone gain. Offering plug-and-play simplicity, the class‑compliant Lyra operates with 24-bit conversion and can run at sample rates of up to 192kHz. No drivers are needed for Mac OS or Windows operation, and for iOS use an Apple Camera Connection Kit USB adaptor will do the trick. AKG specify the frequency range of the mic as 20Hz to 20kHz, though this isn’t as informative as a proper frequency response (-3dB points) figure. The maximum SPL is 129dB, so as long as you don’t use it for close-miking kick drums or trumpets, you probably won’t run out of headroom. I couldn’t find a noise specification but ,in its intended use, noise was never an issue. In use, I found the desk stand to be very solid, and weighty enough at 900g to keep the mic in place. It also has a rubbery non-slip base to further prevent it sliding around. The mic delivers a clean, relatively uncoloured vocal sound but it also proved itself perfectly capable when recording acoustic guitar and a wooden flute. The various patterns are very useful, and although the first option (cardioid) is the most obvious choice for podcasting or recording vocals, those stereo and double-sided

September 2020 / w w w . s o u n d o n s o u n d . c o m

options provide practical alternatives when it comes to recording music, dual voiceovers or capturing a stereo source. Note though that the headphone volume control adjusts both the DAW playback and the mic’s direct monitoring level, so if the balance isn’t to your liking, you’ll need to adjust your DAW’s output level. Some other USB mics provide a DAW/Source balance control, which does make life a little easier. This is a very flexible little microphone that goes well beyond the usual remit as a simple podcaster’s USB mic. The fact that it looks great doesn’t hurt either.  

summary The Lyra sounds clean and quiet, looks fantastic and adds some useful functionality to the usual USB mic format.

££ TT EE WW WW

£139 including VAT. Sound Technology +44 (0)1462 480000 [email protected] www.soundtech.co.uk www.akg.com

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ON TEST

Yamaha THR10 II Guitar Amplifier

Though small and quiet, this stereo combo is big on possibilities...

PAUL WHITE

T

he THR10 II is the smallest in Yamaha’s II series, measuring just 368 x 183 x 140 mm and weighing only 3.2kg. Despite this, and looking like a typical lunchbox amp head, it’s actually a tiny combo that caters not only for electric guitar but also for bass and acoustic guitar. What’s more, it has stereo capability: as well as the digital modelling side of things, it houses a pair of robust 3.1-inch speakers, each driven from its own amp. Yamaha describe the two THR10 II amps (there’s a standard THR10 II and a ‘wireless’ THR10 IIW version, of which more later) as ‘third amplifiers’, for use on occasions when serious sound quality is required but excessive volume isn’t. And judging by the tests I’ve conducted, it could certainly earn its place either in a studio that’s stuck for space, or one in which there’s a need to keep a check on sound levels.

Overview While the name suggests this might be a 10W amp, it is in fact rated at 20W when powered by its included mains adaptor, or 15W when running off a its internal lithium battery. (The latter gives you around five hours

12

September 2020 / w w w . s o u n d o n s o u n d . c o m

of playing time, so if you fancy trying your hand at busking or playing at the bottom of the garden you’re good to go.) Although it’s not monstrously loud, it still packs a hefty punch if you crank it up, and it could even be loud enough for a small pub duo gig if you don’t have to compete with a drummer. For those who are new to the world of recording, both versions come with a download code for a free copy of Steinberg’s Cubase AI (for Mac/Windows) and Cubasis LE (for iOS) DAW software. The THR10 IIW wireless version, which is otherwise identical to the non-wireless version, caters for wireless guitar playing using the optional Line 6 Relay G10T transmitter. It works only with the G10T, not with older Line 6 Relay radio transmitters, and it charges the device for

you as long as the amp is connected to a mains power supply, so you won’t need to purchase a separate charger. All you have to do is plug the Relay G10T into the input jack, which causes it to sync its transmission channel with that of the receiver built into the amplifier. As it works on the 2.4GHz wireless band, the Relay G10T requires no licence. The bottom, rear and sides of the cabinet are made of thick, moulded plastic, but the top panel and speaker grille are fabricated from sheet metal, giving the amp a robust, professional feel. Orange LEDs glow behind the grille, and the power button has an LED ring that also lights orange when powered up. When powered down but still plugged it turns green to indicate that the internal battery is charging. The THR10 II features some inbuilt effects, a choice of amplifier type, and the ability to play back external audio in stereo, for when you get the urge to play along. There’s a USB port for connecting to the free editor app, which comes in two versions: one for Mac/ Windows machines, and a mobile version for both iOS and Android. Usefully, it’s also possible to record and play back directly via USB; the amp appears as

a class-compliant interface. Another DI recording option is the speaker-emulated headphones output — you can connect this to your audio interface’s line input, or feed it into a more powerful amplification system if you need to. (The main speakers are muted when phones are connected.) Should you be looking for more, it’s worth mentioning that the larger THR30 II has a dedicated line output for this purpose. The amp models are all based on Yamaha’s established VCM component-level modelling, and they are controlled using Gain, Treble, Middle and Bass knobs, along with the amp’s Master volume control. Two further controls adjust the overall levels of the guitar amp sound and the external Aux stereo mini-jack or Bluetooth source. Five buttons allow the direct storing and recalling of presets. From the front panel, you can access eight models, including acoustic, bass and flat. But this range is expanded considerably when the app is running, because this provides a choice between Modern, Boutique and Classic versions of the selected amp type. The electric guitar amps are designated Clean, Crunch, Lead, Hi-Gain and Special. Two familiar ‘segment’-style rotary controls are used for selecting and adjusting the modulation effects (Chorus, Flanger, Phaser, Tremolo) and the delay/reverb effects (Echo, Echo/Reverb, Spring Reverb, Hall Reverb), and turning these fully anti-clockwise turns off the effects. A tap-tempo button, normally used to set delay times, activates a very accurate chromatic tuner if held down for a second or two. The app offers you greater control over the effects, though, as well as providing a choice of four appropriate cabinet emulations per amplifier type. In addition it allows access to an internal gate and compressor and allows more alternative reverb types to be selected (Room, Hall, Plate or Spring). The Plate sounds beautifully rich and ‘three-dimensional’. Other app features include setting the guitar DI mode as dry or with effects, choosing an EQ profile for the auxiliary audio input, and giving an indication of the battery’s current charge level. Any of the app-only options, including the three amp variants, can be stored as presets in the THR10 II, so you only need to revisit the app when creating new presets that require these deeper functions.

‘hair’ to take away the sterility makes for convincing chordal tones. Its more rocky tones also stack up well against other good-quality modelling products. The amp deals with bass very effectively at modest levels, yet it’s clean enough and bright enough to work

The companion app gives access to lots of additional functionality, including more cab models and more parameters for each effect.

can balance the contribution of each of the two effects. The headphone output is very usable as a DI feed, complete with stereo effects, and that will be useful to some, but I found I still preferred the experience of putting a mic in front of the amp. It doesn’t have to be turned up loud to sound good over a mic, and you can always add some stereo ambience later or use two mics if you want to capture some of the stereo spread on the effects. So, all in all, the quality and sound is impressive for an amp of this size, and the app and USB interfacing make it really versatile. The THR10 II may look compact and cute, but it’s certainly no toy. Whether you prefer the wireless model or the standard one depends on your requirements and budget, of course. You’d still need to buy a Line 6 Relay G10T to go wireless, but if you do need it, the wireless implementation is very elegant.

“The quality and range of sounds available from this little amp came as a pleasant surprise.”

The Sound Both the quality and range of sounds available from this little amp came as a pleasant surprise. Using the app to navigate the three flavours of each of the amp settings and try different cab models I could get close to just about any of the classic guitar tones. What’s more, you still get the sense of low-end punch that you’d expect from a larger combo, which is impressive from such small speakers. Those on-the-edge sounds that break up as you dig that bit harder into the strings can be set up fairly easily using the Crunch Amp setting, and dialling in a sound that has just a little

with acoustic guitar using the Acc setting. Yamaha also use their own stereo enhancement system, called Extended Stereo, to create very spacious effects and that’s particularly noticeable on the reverbs. The general quality of the effects is excellent, as you’d expect of Yamaha. Editing them only via the front panel limits your options a little, but the app allows for more flexibility, especially when using the reverb/delay setting. As well as adjusting the individual effect settings in greater depth (you get around the same number of controls for each effect as you would on a pedal of the same type), you

summary A surprisingly mature-sounding little combo that has serious studio applications as well as practice, live and mobile uses. The wireless version is easy to use for those who need it.

££ THR10 II £299.99. THR10 IIW (wireless version) £449.99. Prices include VAT.

WW uk.yamaha.com

w w w . s o u n d o n s o u n d . c o m / September 2020

13

ON TEST

Liquidsonics Cinematic Rooms SAM INGLIS

P

opular wisdom has it that algorithmic and convolution reverbs are polar opposites. The one is endlessly flexible, but compromised in authenticity; the other can be spookily realistic but is essentially preset-based, with limited potential for editing. That might have been true a decade ago, but since then, the boundaries have become well and truly blurred. Impulse responses have been appropriated to bolster the realism of algorithmic plug-ins, while convolution technology has become ever more mutable in the hands of some clever designers. One of the clever designers at the forefront of this process has been Matthew Hill of Liquidsonics. His first product, Reverberate, introduced novel modulation possibilities into convolution,

14

Reverb Plug-in

We bask in the reflective glory of a rather special algorithmic reverb generator. and he’s since developed his own Fusion-IR technology to offer much of the editing flexibility we take for granted in algorithmic reverb. Highlights along the way have included Seventh Heaven, a semi-official recreation of the celebrated Bricasti M7 hardware unit, and Lustrous Plates, an open-ended and versatile plate reverb simulation. With Liquidsonics’ latest product, the process has almost reached its logical conclusion, because Cinematic Rooms is very much an algorithmic reverb, albeit one that still incorporates some convolution elements. It also represents a serious attempt to address the needs

September 2020 / w w w . s o u n d o n s o u n d . c o m

of users working in surround, and if your system is capable, multichannel formats up to 7.1.6 are supported. (Pro Tools, for example, only supports channel formats up to 7.1.2, so a workaround is required for 7.1.4 and 7.1.6.) Cinematic Rooms is available for Mac and Windows operating systems, in VST, AU and AAX Native formats. Both the basic and the more expensive Professional edition support all the relevant surround configurations, but the latter provides more presets and, as we’ll see, more options for tailoring your reverbs within a surround mix. An iLok account is required for authorisation. Being mostly

algorithmic, Cinematic Rooms also doesn’t have an enormous IR library, so needs only 200MB drive space.

Known Unknowns Much about Cinematic Rooms’ interface will be familiar to anyone who’s ever used a reasonably sophisticated algorithmic reverb. As is often the case, early reflections are controlled separately from the reverb tail, and many of the controls in both cases are standard. So, for example, the early reflections section includes controls for the Reflectivity of the virtual space, along with the Diffusion of the reflections that are generated and the spacing between them (Size), while the reverb tail parameters include overall Reverb Time and Pre-delay. High-frequency roll-off and modulation within both early reflections and reverb tail are also controllable in the normal way, and there’s an entirely conventional EQ section. At the same time, though, you’ll also notice a few more unusual parameters. The early reflections section is notable for a control labelled Proximity, offering a variety of settings starting with Adjacent at one end to Far and then Reverse at the other. This idea is to offer more realistic and nuanced control over the apparent position of the source within the virtual space than is obtainable with just a pre-delay control, and the results fully justify the means. The reverb tail section, meanwhile, includes a parameter named Bloom, which shapes the build-up or onset of the reverb.

Plane Simple One of the things that makes Cinematic Rooms unique is a set of parameters available only in the Professional version, where both early reflections and reverb tail feature an X-Feed section. The controls found here determine the way in which reverb from a source in one channel propagates through the others. The available parameters are Level, Delay and Roll-off; the function of each is self-explanatory, but their purpose might not be. The point is to be able to specify the extent to which reverb ‘belongs to’ a source and reflects that source’s position in the sound stage. The value of this feature is arguably most important in post-production, especially in surround sound. Imagine, for example, a film scene where something is moving towards the viewer along a tunnel: the reverb from the

Decorrelation Coding Perhaps the simplest way to create a sense of space around a signal would be to send it to two very short delay lines with slightly different delay times, and pan those hard left and right. If that’s all you did, it would probably be reasonably effective — until someone listened to it in mono. Because the left and right channels in such a delay are identical apart from a few milliseconds’ time difference, collapsing them together will cause them to interfere with each other and introduce unpleasant comb filtering. This is an extreme example of ‘correlation’, where two signals have ongoing similarities that bring about audible consequences when they’re combined. There are various things that can be done to reduce the level of correlation between two related signals; in the example above, the use of slightly different pitch-shift settings on either side is a well-known trick. In a well-designed reverb plug-in, every channel’s output should be fully decorrelated from that of every other channel and from the source itself — whilst, of course, retaining enough of the source’s sonic character to be recognisable as reverberation from that source. This is true of Cinematic Rooms and it’s one

tunnel would be very pronounced, but also limited to the side from which the source is approaching. At the other end of the spectrum, completely decoupling the reverb propagation from the source position could also be an effective tool in scenes where an actor is struggling to locate the origin of a sound. This brings us to the other main distinction between the basic and Professional versions of Cinematic Rooms, which is a concept Liquidsonics call ‘reverb planes’. As I’ve already mentioned, the basic version supports all feasible surround formats, but with the limitation that all settings apply to all channels equally. There are plenty of circumstances where that would be more than adequate, and anyone using Cinematic Rooms solely as a stereo reverb will probably get by fine without the Professional version. But if you’re mixing for surround configurations such as Dolby Atmos that offer precise localisation in all three dimensions, there are also circumstances where you want things to behave in different ways in different planes. Consider, for instance, how sound might behave in a long, thin hall with no roof. Apart from a few early reflections from the floor, there would be no reverberation in the vertical plane, while the left-right ambience might have

of the reasons why it still sounds very good even when auditioned in mono. But what no reverb can do is eliminate correlation between the source signals that are being fed to it. In other words, if you took those two delay lines and sent both of them to the same reverb plug-in, the resulting reverb would inevitably sound comb filtered regardless of the quality of the plug-in. Cinematic Rooms Professional introduces a novel way around this. If you want to apply the same reverb setting to two or more sources in your mix that are correlated, you can set up multiple instances of the plug-in and choose different ‘decorrelation coding’ patterns for each. In the example above, you could apply the same hall or chamber preset to each of the delay lines, and as long as you used different decorrelation coding settings and had Cinematic Rooms 100-percent wet, you’d hear no comb filtering. In truth, I struggled to find a real-world application for this feature — ultimately, if your source tracks don’t combine unproblematically in mono then you have bigger problems than correlated reverb! — but it’s a thoughtful provision.

a significantly different character from the front-back reflections. And in Cinematic Rooms Professional, this sort of effect can be recreated. All parameters default to global control, but they operate within individual stereo ‘planes’: left-right, rear left-right and so on. Any parameter for any plane can be de-linked from the global setting and varied: so, in the example above, the horizontal planes

Liquidsonics Cinematic Rooms $199/$399 pros • Sounds great. • Unusually detailed control over early reflections. • The concept of ‘reverb planes’ will be an asset to anyone working in surround, especially in post-production.

cons • If you want trashy reverbs, you won’t find them here.

summary Cinematic Rooms is a very high-class algorithmic surround reverb, and although some of its features are designed with post-production users in mind, it works extremely well in music mixing and in stereo too.

w w w . s o u n d o n s o u n d . c o m / September 2020

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ON TEST L I Q U I D S O N I C S C I N E M AT I C R O O M S

might have similar settings for things like Reflectivity and Proximity, but varied reverb times and balances between early reflections and reverb tail, while the vertical planes might be configured very differently. The individual planes are represented by discs that appear at the top right of the user inferface, each displaying a number representing the number of controls that are unlinked within that plane. There are a few other features that are available only in Cinematic Rooms Professional. Perhaps the most significant is that Professional users get more than 300 presets to the 80 or so in the basic version, but there are also a number of extra controls. The tail algorithm features a built-in delay line and an intriguing control labelled Undulation; the action of this parameter is somewhat obscure, but quite noticeable in long tails, where it can create a subtle animation and movement that’s very different from conventional LFO-derived modulation. The early reflections’ Diffusion is augmented in the Professional version by an additional Width parameter, and the Proximity control is joined by a setting labelled Pattern. This adds further versatility to the options for spacing the reflections, including an interesting Nonlin U setting. In conjunction with the various delay and Proximity settings, Nonlin U permits the creation of reverbs that behave in a ‘nonlinear’ fashion, with reversed or flattened dynamic envelopes, yet still sound plausibly natural. I’m not sure I’ve ever come across another reverb that can pull off this particular trick!

In Use The concept of reverb planes is a deep one and hugely powerful, but I think it’s fair to say that it will be of most interest to those working in high channel-count surround formats such as Atmos or higher-order Ambisonics. That probably doesn’t include most people who are recording and mixing music, so does that mean this plug-in isn’t

Alternatives For stereo use, high-quality algorithmic reverb plug-ins are available from the likes of Lexicon, Relab, FabFilter, Sonnox and many more developers. Surround reverbs are less common; I’m not aware of any rivals that have all the same features as Cinematic Rooms but Flux’s IRCAM Verb v3 probably comes the closest. Exponential Audio’s reverbs are also surround-capable and rightly popular in post-production circles.

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Reverb planes in action: the dials at the top right of the screen indicate that three parameters in the front plane have been delinked from the Master settings, two in the Rear plane and two in the Centre plane, which is currently being edited.

relevant for music users? Absolutely not. Cinematic Rooms is a hugely powerful and great-sounding algorithmic reverb for any sort of mixing, music included. Sound quality is straight out of the top drawer, and what’s really noticeable in use is the attention that’s been paid to getting the early reflections right. Early reflections are the most important auditory cues as to the size and shape of a real space, and Cinematic Rooms lets you dive unusually deeply into their simulation without introducing millions of parameters. One upshot of this is that the plug-in especially shines at producing ambiences and other short reverbs. Where lesser plug-ins start to ‘bark’ or become unnatural and metallic-sounding, Cinematic Rooms just adds ever more delicate air or richness to the source. If you like reverbs of the “subtle enhancement that no-one will notice it until it’s bypassed” variety, you’ll be spoiled for choice here. That’s not to say that Cinematic Rooms can’t do more obvious reverb too, of course. With a maximum reverb time of 45 seconds, plus the ability to do infinite reverbs if you want, it has no problem recreating both plausible concert spaces and epic Valhalla-like halls. The Professional edition includes a large number of realistic post-production-oriented rooms and other interiors, along with lots more impressionistic smallish spaces for music production. The Chambers category is

September 2020 / w w w . s o u n d o n s o u n d . c o m

a particularly rich source of inspiration for music mixes, and I’m sure very few real reverb chambers ever sounded half as sweet as presets like ‘Helios Chamber’. Even settings that really ought not to work, and almost certainly wouldn’t in other plug-ins, deliver results that are not only pleasant but useful. I would never normally think to use a nonlinear reverb on a lead vocal unless as a special effect, but there are several here that work beautifully. And as well as offering levels of surround controllability rarely encountered, Cinematic Rooms also sounds great in mono, which is an important quality in music mixing. The only conceivable downside is that its absolute refusal ever to sound bad might not always be exactly what you want in a reverb. Trashy, gritty, grainy, metallic, ringing and obviously artificial reverbs all have their place on occasion, and I never managed to push Cinematic Rooms into producing anything that could be described in those terms. Settings that ought to sound all wrong just, well, don’t! If lo-fi is your thing, buy something else; but if you want to experience a reverb that can make any setting at all sound hi-fi, Cinematic Rooms will deliver every time. ££ Professional version $399; standard version $199.

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ON TEST

Cranborne Audio EC2

Dual-channel Mic Preamp With added functionality and improved specs, this great preamp just got better.

HUGH ROBJOHNS

W

hen it comes to choosing mic preamps, some value clean performance, while for others ‘character’ is all. A couple of years ago, the clever people at Cranborne Audio came up with a way of delivering both in one design, with a very clever, controllable means of adding different characters that they called ‘Mojo’. It was introduced in the superb Camden 500 preamp (reviewed in October 2018: go to https://sosm.ag/ cranborne-camden500 to read more), which impressed the editors at SOS so much it was Highly Commended in our Gear Of The Year reviews. Cranborne have recently built on this preamp design to create a new 1U rackmounting device, the EC2. The name is not an expensive London postcode, but rather it derives from ‘Expander

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Camden 2-channel’ and it is so described because the EC2 combines a pair of Camden preamps with two high-quality headphone amps and Cranborne’s Ethernet-based CAST interface system. The CAST system and headphone amp design were introduced in the company’s 500R8 500-series interface (reviewed in September 2019: https://sosm.ag/ cranborne-500r8), and used in the 500ADAT (reviewed November 2019: https://sosm.ag/cranborne-500ADAT). As the EC2’s various individual elements have already been covered in previous reviews, I’ll focus here mostly on how the EC2 works as a complete package.

Preamp Overview Although the preamps in the EC2 appear much like the Camden 500 module turned on their sides, and sound every bit as good, the construction is very different.

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All the electronics for both preamps, the headphone amps, and the power supplies sit on a large motherboard. Subsidiary cards carry the front-panel controls and rear-panel connectors. Surface-mount components are used throughout, and an elaborate DC-DC converter section generates the internal power rails from an external 24V DC line-lump PSU, which connects via a locking ‘Power-DIN’ plug and has a standard IEC mains inlet. Unusually, the mains safety earth is also extended to the rack chassis via the negative side of the DC supply, and a marked screw on the rear panel is also provided as a chassis grounding terminal, if required. An unplanned benefit of laying out the Camden preamp circuitry for the EC2’s much larger circuit board was slightly improved intermodulation (IMD) and total harmonic (THD) distortion figures, both

Cranborne’s clever Mojo circuit can transform this brilliantly clean, quiet preamp into a wonderfully characterful colour machine.

of which were already spectacularly good! Another small improvement, introduced in the Camden 500 module after serial number 500 and carried over into the EC2, improved the rejection of RF interference. Operationally, the preamp sections have the same controls and layout as the Camden 500: a switched gain control, with 12 5.5dB steps, avoids gain-bunching problems and allows accurate matching of channels; there are toggles for polarity inversion, a high-pass filter (80Hz, first order), phantom power (with status LED) and input source (mic, line or DI). The Mojo section has another toggle to select its Cream or Thump modes and a rotary control for the amount, with a backstop switch to turn the facility off. A single multicolour LED indicates the signal level.

On the rear, each preamp has a single XLR for the mic input, with a toggle switch to select the remote CAST input instead (see below). On the front, a quarter-inch TRS socket caters for balanced line and unbalanced instrument DI connections. The preamp outputs are presented on both XLR and TRS sockets, and a ground-lift toggle completely isolates pin 1 of the XLR (but doesn’t affect the sleeve ground of the TRS socket) when engaged. A brand‑new facility for the EC2 is a TRS output socket labelled ‘Link’, which provides an actively buffered, impedance-balanced output right after the input selector, and before the gain stage and Mojo sections. This is intended to feed an instrument amplifier when using the instrument input, like a DI box, but can also be used for re-amping duties when using a line input. It’s a very thoughtful, handy addition.

Headphone Amps The EC2’s headphone amps are reassuringly quiet and clean, and they’re very powerful, being able to deliver over 0.5W into each earpiece for headphone impedances between 32 and 220 Ω, the power delivery being at a maximum for headphones around 100Ω impedance, reaching a chunky 1.21W. I could drive all of my headphones (ranging from 32-250 Ω) to painfully loud levels without difficulty. Each headphone amp has a front-panel quarter-inch TRS output, and the output volume is adjusted with a slim rotary knob. The other three knobs set the input source levels, with one for each preamp channel

and one for an external stereo aux input. The long, thin control knobs look and feel good, but I’d have preferred some visual distinction for the master level controls. The external auxiliary input comes either from a pair of rear-panel TRS ‘Aux’ sockets, or the ‘CAST Out’ return signal. The Aux sockets take priority, and plugging only into the left input provides a dual-mono signal. By default, the two preamp signals are fed to the headphones in stereo, with preamp 1 in the left ear and preamp 2 in the right, but a front-panel

Cranborne Audio EC2 £1099 pros • A pair of superbly clean and quiet Camden preamps in a 1U rack box. • Mojo circuitry is astonishingly characterful, versatile and controllable. • Two independent, powerful headphone amps with external cue inputs and full mixing facilities. • CAST interfaces for low-cost stagebox and single-cable interconnection options. • Very nicely engineered.

cons • The line-lump PSU will disappoint some potential customers.

summary Cranborne have repackaged their brilliant Camden preamps into a dual-channel rackmount format, combined with a pair of powerful independent headphone amps to provide convenient and flexible latency-free monitoring.

w w w . s o u n d o n s o u n d . c o m / September 2020

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ON TEST CRANBORNE AUDIO EC2

As well as the expected inputs and outputs, the rear panel plays host to the CAST ports and a handy ‘Link’ socket, which allows you to pass a ‘thru’ signal from the instrument input to your amp, or to re-amp mic/line signals.

switch allows auditioning in mono (both preamps in both ears). Not many two-channel mic preamps include headphone amps, but it makes a lot of sense for modern project-studio workflows and practices. For example, when working out the best position for placing a mic relative to an acoustic instrument it’s essential to be able to listen as the mic is moved around; being able to plug headphones straight into the preamp to do that is remarkably convenient. But the real raison-d’être is to provide convenient, high-quality, latency-free, independent monitoring for up to two performers while recording. Each artist can listen directly to their mic, with or without a contribution from the second mic/instrument input channel, and with a mono or stereo backing track via the external inputs. The artist can also balance the input source levels and overall volume exactly as required from the front panel. In a small project studio or on location, these facilities make life so much simpler and easier.

CAST CAST is an abbreviation of ‘Cat5 Analogue Signal Transport’ and is Cranborne’s method for using low-cost shielded Ethernet cabling instead of traditional analogue multicore. The system uses

Alternatives Audio interfaces aside, there aren’t many dual-channel preamps with headphone monitoring. The Neve 1073DPX and the new Neumann V402 both have a single headphone amp to monitor the preamp outputs, but neither accept external cue mix inputs. The only preamp that immediately comes to mind with headphone monitoring and a cue mix input is the Focusrite ISA One, but that is only a single mic preamp with a separate DI channel on the side.

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the four twisted pairs and screen of an Ethernet cable (of up to 100m) to convey four balanced audio signals, two in each direction. Although Cat5 is mentioned in the name, shielded Cat5e, Cat6 or Cat7 cables can be used, and individually shielded Cat7 is preferred when working with mic-level signals. It’s an entirely passive system — in the breakout boxes the RJ45 sockets are wired directly to the XLR/jack sockets — so it’s a very cost-effective and convenient way of extending connections. On the back of the EC2 are two CAST sockets, one labelled ‘CAST In’ and one ‘CAST Out’, although both actually carry two channels in each direction. The CAST Out socket carries the analogue line outputs from the two preamp channels, while its return signals are routed to the headphone amps, along with the stereo auxiliary inputs. A typical application for this would be to connect an EC2 preamp directly to a 500ADAT or 500USB rack, for example, using just the one Ethernet cable, with a monitor mix from the rack being sent back to the EC2’s headphone amps. On the input side, the CAST In socket allows remote input connections to the preamps, so an Ethernet cable can be used to link the EC2 to a CAST breakout box elsewhere in the room, serving as a snake and stagebox. The two return signals sent back down to the CAST breakout box come from the headphone monitor mixers of the two headphone amplifiers, at line level. The longest shielded Ethernet cable I had to hand was only five metres, and the interface seemed to work perfectly well with line, instrument and mic-level sources over that distance, with any crosstalk between channels completely inaudible.

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However, for longer runs, and especially with mic-level signals, I’d probably want to use a Cat7 cable.

Verdict I was hugely impressed with the sound quality, technical performance and versatility of the Camden 500 preamp, and the EC2 version is every bit as good; it remains one of my all-time favourite preamps. It boasts not only superbly clean, quiet and dynamic performance as a straight preamp, but also an extraordinary ability, courtesy of the Mojo circuit, to become the most versatile, characterful preamp I know. The fact that the EC2 has slightly better technical specifications than the Camden 500 is a pleasant bonus, although I doubt anyone will hear the improvements unless working in an area with strong RF interference. The rackmount format actually makes it more desirable to me, and the inclusion of headphone amps that match the high-quality standards of the preamps opens a new world of convenience and practicality. I could see many more manufacturers following this idea in the future. Overall, I have to say that the EC2 is one of the most desirable, capable, versatile and practical preamps I’ve used in a very long time. Given its phenomenal audio quality and flexibility, with two excellent preamps, two headphone amps, and the CAST interfacing options, it represents exceptional value for money, too. If you’re in the market for a serious new preamp, you’d be mad not to put this at the top of your shortlist. ££ £1099 including VAT. EE [email protected] WW www.soundtech.co.uk WW www.cranborne-audio.com

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ON TEST

DDMF Magic Death Eye Compressor Plug-ins

These two plug-ins bring you the spirit of their ‘unobtainium’ hardware counterparts! ERIC JAMES

I

first heard of the Magic Death Eye analogue compressor from an American mastering colleague who’d just taken delivery of one. Intrigued, I looked for more details but sadly soon found a few good reasons for me not to rush out and get one. First, designer/ builder Ian Sefchick can only produce them when free from the demands of his mastering career at Capitol Records, so his output is both limited and irregular. Second, even if one were to become available, he doesn’t make them for export outside the USA. Finally, the stereo version costs $8,800! That’s actually not an

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unreasonable amount to ask for a superbly designed, hand-built vari-mu compressor, but it is hardly chump change. So when I learned that DDMF had created a plug-in version, I quickly tried the demo. It was good, and within a couple of hours I’d bought it. Recently, DDMF released another ‘stereo’ Magic Death Eye plug-in. I bought that one immediately too and both plug-ins now have a home in my mastering workflow and have already been used on several professional projects. There are versions for the common Mac OS (v10.8 or higher) and Windows (32- and 64-bit, v7 or higher) plug-in formats, and an AUv3 version for iOS. Installation and activation on my Windows

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10 and Sequoia 14 system was blessedly simple: the license is provided in a ZIP file, and you locate it when you first use the plug-in. Sadly, I’ve no way to confirm how close the sound of the plug-ins are to the originals, so I’ve had to assume a very close kinship and evaluate them on their own terms.

It Takes Two Despite the names, both Magic Death Eye plug-ins are fully stereo in operation. One, which I’ll refer to as MDE, is called ‘Mono’ because it is based on the mono version of the hardware; the ‘Stereo’ version (MDEST) is modelled after the stereo hardware unit. There are enough differences between the plug-ins to consider having both in the toolbox, though their basic design is similar and you’d be unlikely to use them in the same chain. Both have variable ‘switched’ input and threshold level controls, six fixed time-constant parameters (labelled 1 to 6, as on the Fairchilds that inspired the hardware MDEs) and fixed-frequency, switchable high-pass side-chain filters (set at 150Hz for the MDE; and at 200Hz for the MDEST). The MDEST has an additional Master output level knob, whereas the MDE’s is hidden, and the MDEST has a ‘Threshold Link’ (which seems a bit

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confusing at first, as a single Threshold control applies to both channels, but in practice it does tighten things a bit). Both models also have a GUI which is a painstakingly accurate representation of their respective hardware. The screw adjusters of the analogue originals don’t all have a corresponding digital function, but some, along with the Magic Death Eye faceplate logo and the power lamp on the MDEST, have hidden software-only features. On the MDE, the logo switches in a ‘Punch’ mode (which changes the compression characteristics and adds a new graphic to the GR meter), and the screws between the input and threshold controls adjust Output level (-20 to +12 dB) and wet-dry mix (0 to 100 percent). For the MDEST, the MDE logo switches the upsampling rate from normal (Black) to x2 (light blue) to x4 (dark blue). The power lamp brings in the same Punch mode (now known as ‘Laura’, and indicated by the L meter label changing to the name). This time, the balance control screws adjust the harmonic content from the normal 50 percent (status lights orange) to 0 percent (green) and 100 percent (red). At present there’s no wet/dry mix control on the MDEST. The most significant difference between the two versions is that MDEST has a simple but interesting EQ section, which operates within the compression circuit. This includes a low-cut filter (off, 20 and 30 Hz), a switched LF boost (20, 40 and 100 Hz, with up to 5dB of boost in 1dB steps), and a switched HF boost (5, 12, and 18 kHz, also with 5dB of boost in 1dB steps). Right-clicking on the Low frequency control changes the low end to an elliptical filter, mono’d below the chosen frequency, and right-clicking on the High frequency control switches the EQ path from stereo to Mid or Sides.

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summary In attempting to model two of the very rare Magic Death Eye compressors, DDMF have created some special plug-ins.

& MA

cons

Chris Korff, Sound On Sound

D

NED, DEV IG

OPED EL

• Both sound great, and are pretty difficult to make sound bad! • Useful ‘hidden’ functions. • Discounted bundle pricing.

Smooth sound, ‘‘ built to last… I really can’t fault it!’’

K

DDMF Magic Death Eye

Casey Cohen, Podcast/Video Producer

DE S

To my mind there are basically two kinds of mastering processor: one has a complex but very flexible set of parameters, which you have to think your way through; the other, by reason of simplicity and lack of parameter information, you can only learn by listening. The MDE plug-ins are analogue simple, so I approached evaluation of both in much the same way as the hardware Hendy Amps Michelangelo I reviewed earlier this year (go to https://sosm. ag/hendy-michaelangelo to read more). I’ll start with the original MDE. As the ratio is not separately adjustable, there are only really four parameters which you can use to change the character of compression: the input level, the threshold, the six preset time

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w w w . s o u n d o n s o u n d . c o m / September 2020

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ON TEST D D M F M A G I C D E AT H E Y E

Clicking the logo engages Punch mode, which transforms the compressor behaviour in a wonderful way.

parameters, and a choice of three additional attack settings (fast, medium and slow). Getting things right is a matter of listening and tweaking, but on an attended jazz stem mastering project, with a lumpy, bumpily recorded acoustic bass and a female vocalist with a nicely textured mezzo voice, I found that I could achieve the desired effects pretty quickly. It was interesting to get the immediate responses of the musicians: the bass player thought his tone benefitted from a slight push on the input levels, and for most of the songs a medium attack and setting 3 or 4 seemed to do the trick, producing an articulate snap when needed and controlling any boom. The vocalist also liked the medium attack, with setting 3 sounding the most natural for ballads, but Fast and setting 1 preferable for up-tempo and scat singing. When I started mastering stereo mixes, it was quite easy to ‘over-control’ things, even when using the classic combination of a slow attack and a relatively fast release (eg. setting 2). It seemed to grab just a bit too much, pushing middle images to the back a bit and dulling things down. Dialling back on the threshold to the extent that the gain reduction needles never even moved was a revelation. With such a minimal setting (and the compressor likely doing a bit more than the GR needles indicated) the MDE seemed to dance along with certain kinds of music, adding a seductive movement. I wasn’t always entirely happy with the effect on the kick when mastering certain styles of music, even with the 150Hz side-chain high-pass filter engaged. But

Alternatives There are many good modelled vari-mu plug-ins available, including the Pulsar Mu and the Kush Audio AR-1 and various Fairchild emulations, but there are none that offer quite the same character as DDMF’s Magic Death Eyes.

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then I discovered a ‘hidden’ feature, not even mentioned in the early versions of the manual: clicking on the Magic Death Eye logo changes the background of the meter to the kind of pin-up that used to feature on the nose cones of American World War II bombers, and does something quite startling to the compression characteristic. It’s called the Punch mode, and it is very nice. It allows much more kick to come through and overall gives the sound an additional punch and push. (I used this on a track for a forthcoming Faithless album I was just finishing off at the time.) MDEST is clearly of the same bloodline, but it has been optimised for mastering. The compression is much more gentle, the high-pass filter has an extra 50Hz setting, and then there’s the EQ. Putting this plug-in in the processing chain for the first time brought a very familiar change to the music running through it. There wasn’t much of a feeling of restraint, but there was a very nice additional density. Glue can become gloop very quickly with some analogue vari-mu compressors I’ve used in the past, though. So when pushing things a little too hard and still liking the density but not the darkness, the hidden feature that allowed me to dial down some of the tube characteristics worked very well. The compressor action with the harmonic distortion reduced to about 30 percent is very seductive, and I’ve now used it on a lot of mastering projects. The EQ section can add some lovely touches but needs a lot of careful listening. Although the manual reports them as a relatively steep 24dB/octave, the 20 and 30 Hz HP filters seemed to reach an unexpectedly long way into the lows, so sometimes I used the 20Hz one but I didn’t use the 30Hz one much at all. I didn’t use the low-boost, narrow bells at 20 and 40 much either, but that’s simply because I had them just prior to an analogue chain, which was often the wrong place to use them. In a different setup, the 100Hz shelf at 1dB can add some nice juiciness to

September 2020 / w w w . s o u n d o n s o u n d . c o m

hard-sounding bass. The medium-wide HF bell boost got rather more use (in mastering, almost all at 1dB), as 12kHz was occasionally really nice for adding a touch of percussive clarity, especially when used on the Sides signal only. This added width without sounding like an artificial enhancement, and the 18kHz added enough gentle air that I sometimes used it despite having a nice analogue EQ, since it freed up an upper band on the latter. At high input levels, it’s possible to induce some controlled clipping, as indicated by the GR meters starting to flash red. Judging the amount is made easier by the ability to link the input and output controls (by ctrl-clicking one of them) so that you get a relatively unchanging monitoring level, though a higher input will also need a more moderate threshold. All of this allows some additional perceived level, and it works pretty well for that.

The Eyes Have It! Hopefully, this review makes it obvious that I’m very taken with both versions of DDMF’s Magic Death Eye. As a mastering engineer, if I could really only have one I’d choose the MDEST, whereas if wanting something for use on the stereo bus while mixing, I’d more likely choose the MDE. That said, there are some things in mastering that still seem to be to be done better with the MDE, and here’s just one example. In the latest update to the MDEST, a Punch mode has been added. I didn’t miss it previously, because I could and always did use the MDE for that. But even now, the two modes are not directly comparable, and a lot of the time I prefer the ‘bigger’ punch of the MDE. It’s just as well, then, that there’s a discount available that makes the prospect of buying both plug-ins look rather more attractive! ££ MDE £79. MDEST £119. MDE Bundle (both versions) £159. Prices include VAT.

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September 2020 / w w w . s o u n d o n s o u n d . c o m

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INTERVIEW

KAITLYN AURELIA SMITH

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September 2020 / w w w . s o u n d o n s o u n d . c o m

We talk to prolific producer Kaitlyn Aurelia Smith about her ambient compositions and love of Buchla synths. TOM DOYLE

I

n the increasingly crowded field of electronic ambient composers, Kaitlyn Aurelia Smith stands out. Working with modular synths — mainly Buchla systems — Smith’s meditative, new age-y music was initially inspired by ‘60s minimalist pioneer Terry Riley, but has developed since 2012 over eight albums, resulting in the dreamy soundscapes of her latest, The Mosaic Of Transformation. Growing up on Orcas Island, off the Pacific Northwest coast of America, Smith was first motivated to make music at the age of 15, after watching a friend of her family playing piano during a visit to their home. “I remember seeing the joy that it brought them,” she recalls. “I could feel what it felt like to play the piano. After they’d left, I remember sitting down at the piano and starting to try and emulate what I saw in them. Of course, it didn’t sound like what I heard for a while [laughs]. But it sparked kind of like a muscle memory in me that I didn’t know was there.” In her late teens, she turned to classical guitar, going on to study sound engineering, orchestration and composition at Berklee College of Music in Boston. On a visit home, she happened to mention to a neighbour that she was a Terry Riley fan. By chance, the neighbour owned a ‘60s-built Buchla 100, which they generously loaned to Smith for a year. “That kind of reopened music for me in this lovely pressure-free way,” she says. “It was a really magical time in my life. I felt like the Buchla was a perfect way to explore sound in a really new way for me. Because even though I’d just come from music school, the aspect of zooming in on what makes sound and how much can be heard in one note I felt was not really covered in music school. It’s one of those topics that people just assume, if you’re into music, it must be innate knowledge. But there’s so much wisdom within that basic foundation of sound.” Initially, Smith experimented by feeding her guitar through the Buchla

and treating its sound. Then, progressing past making what she calls “bleepy bloop” sounds, she’d work with two oscillators and make incremental changes. “Mostly just listening,” she explains. “I would practise just making really subtle adjustments and trying to understand what was happening to the sound. Trying to just really break into how many things change when you make a subtle adjustment. I had just come from intensive study of building on sound, and I kind of wanted to go the reverse way of deducing how sound is affected. So, it felt like this research project in the beginning. “Just in general, I’m always wanting to find, I guess for lack of better language, ‘beautiful’ sounds in every instrument. And for me beautiful sounds aren’t just the pleasing sounds. It’s where two extremes are meeting, or where two things that don’t feel like they have something in common are meeting. So, I think that’s why it was really fun for me to explore with the guitar and the synthesizer. Trying to find how they can connect through their capacity to, like, move air.”

Smith stresses that, like the original Music Easels, the new models all have their own individual characters. “I’ve played, like, four or five of the newer ones and they all are still very different, even though they’re remade. Honestly, all Buchla instruments are really different. A lot of the original Buchla components aren’t being made any more or are harder to find. So, as technology progresses and older components get discontinued, it changes the sound and it changes the way that it interacts.” While she sometimes uses Moogs in her recordings, Smith is very clearly devoted to Buchla synthesizers. “Well, they’re different kinds of synthesis,” she points out. “Buchla synthesizers are additive synthesis and Moog synthesizers are subtractive, so they have different approaches.” Smith’s third and fourth albums, 2014’s Tides and 2015’s Euclid, were made exclusively using the Music Easel. “As far as the sound goes,” she says, “for me there’s a certain resonance in Buchla instruments that I really connect to. To me, it has a more potent tone.”

Easel Does It

In terms of DAWs, Kaitlyn Aurelia Smith tried out various ones before settling on Ableton Live. “At school I had learned Logic, DP and Pro Tools,” she says. “So, Ableton was a really new world to me. I learned it six, seven years ago. I just like how seamless it is to go between recording and the live performance. It just seems like you can create this environment in the software so that it can easily translate to live. ‘Cause I really like to be in control of mixing my own sound. So, when I perform, I always use Ableton so that I can set up my effects in it, third party or not. I can keep my equipment load pretty light for travelling and just have the hardware be the synthesizers and not worry about bringing a lot of pedals and other things.” On her fifth album, 2016’s Ears, Smith broadened the range of the synths she used, to include the EMS Synthi, ARP 2600, OSC OSCar, Korg Mono/ Poly, EML ElectroComp 101 and Moog Werkstatt-01. When it came to the rarer, older or more expensive items of gear, she travelled to various studios or company HQs where the synths were in residence and she could capture their sounds on the fly. “I started really learning about rare vintage synthesizers,” she says. “I would

Kaitlyn Aurelia Smith’s big discovery, however, and one that was to help shape her sound, was the Buchla Music Easel. In 2013, Smith bought one of the new Easels developed by BEMI and launched only three years before Don Buchla’s death in 2016. “I bought my own,” she says, “or I guess I did with the help of a community. It was kind of a funny story because my husband and I asked for donations at our wedding for us to buy a cow [laughs]. And then we ended up moving to Bolinas, which is a small town in West Marin, California, and we didn’t have space for a cow there. So we used that money for a Music Easel instead. “This was one of the first remakes,” she adds. “We were on a wait list before they had really shared that they were remaking the Easel. This was when Don was still a part of the company. We were number 15 on the wait list and then we got an email that said that the first three or four were ready, but they didn’t have these certain nuts that screw onto the switches. They asked if there was anyone who wanted it, even if it didn’t have those. And we were like, ‘Yeah, we want it now.’ And so, we got bumped up.”

Tools Of The Trade

w w w . s o u n d o n s o u n d . c o m / September 2020

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INTERVIEW K A I T LY N A U R E L I A S M I T H

The Buchla Easel is at the heart of all of Smith’s live and studio productions.

find out where they still exist and visit them and just record everything I did. So, I would just have all these samples or lots of sessions of recorded takes that I didn’t know what I was gonna do with it yet. “That kind of began a study process for me of seeking out residencies in places that have this equipment, rather than trying to own it. ‘Cause it’s very expensive and it’s a bigger responsibility. But they all have really incredible aspects. I especially appreciate the tactile interface of the EMS that makes you feel like you’re playing Battleship as you’re patching [laughs].” Ears was the first album that Smith made while consciously keeping in mind the fact that she would have to tour and perform the album live. “That one I wrote just by figuring out what my live set was and then recording it,” she says. “I was practising a live set and then I just pressed ‘record’, basically.”

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In a 2016 ‘In The Studio’ video still available on YouTube, Kaitlyn Aurelia Smith can be seen showcasing some of her techniques, utilising an impressive setup involving both new and old modular gear. In the Eurorack department, she operates Make Noise’s Tempi six-channel polyphonic module as a master clock to trigger various units such as the MATHS and RxMx synth modules and the 4ms Spectral Multiband Resonator. In addition, she uses the ‘70s-era Oberheim Two-Voice Pro and the Studio Electronics Sensei Hybrid System. “I don’t actually own any of those,” she laughs. “Those I was borrowing. The way that I go about equipment in general is either through residencies or finding really kind people to lend me their stuff. Honestly, I don’t have that much experience with Eurorack besides the Make Noise system which I used on the

September 2020 / w w w . s o u n d o n s o u n d . c o m

album The Kid [2017], and some Jomox modules. But that’s about as far as it reaches out for me. “In that video I was lent some Eurorack gear, but it never really stuck for me. I don’t have negative things to say about it, but I just haven’t found a Eurorack oscillator that feels the same to me as the vintage oscillators. “Sometimes I wish I had a set way that I work,” she says. “But I really don’t. It changes from song to song, from day to day.”

Current Rig These days, Smith’s current setup comprises her trusty Music Easel, a Buchla 200e, a Sequential Circuits Prophet-5, an EDP Wasp, a Roland SH-101, Moog Grandmother and an Oberheim SEM. Also, despite her mixed feelings in regard to Eurorack gear, she does use a fair few

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INTERVIEW K A I T LY N A U R E L I A S M I T H

Smith’s Eurorack setup.

modules, including Make Noise’s Pressure Points touchplate controller, Mannequins’ Just Friends tone generator and ALM Busy Circuits Akemie’s Taiko drum voice FM synth. “I feel very fulfilled and like I can make lots of music with all of that,” she says. “And when I feel like I don’t have anything that is within a palette that I’m trying to communicate, then I have a list of residencies to go to.” In terms of sequencers, Smith’s workhorse is her Buchla’s 252e Polyphonic Rhythm Generator, visually distinctive for the interface’s 11 concentric rings displaying the generated patterns. The 252e helps her create the complex polyrhythms that are a central feature of her sound and which come back to her love of Terry Riley. “I’ve always had an appreciation for polyrhythms,” she says. “That’s always felt like the most natural expression of rhythm for me. I love the way that Terry Riley’s melodies and his rhythms kind of float over each other in this very expansive way,

34

where it doesn’t feel like it’s repetitive. It felt like there was always variation happening in the way that it lined up. “That really spoke to me. Even though, to some people, they think that’s really repetitive. Like, I know a lot of people think minimalistic music is really repetitive. But it’s those subtle rhythmic adjustments that have always been really exciting to me. “So, I’m using the 252 and then I do a lot of looping. Looping is oftentimes the way I go about sequencing things. Sometimes I use sequencers, like I have some MIDI controllers that are extensions that have sequencers. But, again, it really depends on what I’m doing.”

Performing From Ears on, Kaitlyn Aurelia Smith has always remained conscious when making records of how she’ll perform her modular music live. “I do have one thing that is like a constant in my workflow,” she says, “which is when I make an album and I know I’m going to perform with it, then

September 2020 / w w w . s o u n d o n s o u n d . c o m

I have a [different] setup for that. I practise for usually, like, a year to a year and a half, and I don’t change that live setup for the whole album cycle. ‘Cause it takes me so long to figure out a live setup.” Nowadays, the centre of her live show is her Buchla 200e. “The 200e has a preset manager,” she says, “so I spent the first four months just programming all of the sequences or all of the presets, ‘cause it takes a really long time to do that. I programmed, like, 17 presets that I’m using in my live set. Then I usually go through a memory process after I programme something, where I just test my memory and my muscle memory. Like, I make a list of how many things can go wrong in the live set [laughs]. And then I come up with a solution for all of them. I practise quickly solving all of those, and I practise unpatching and patching the modular within a certain time period. Then I go through a muscle memory for tuning, where if one of the oscillators were to go out of tune slightly, I have it memorised what fundamental it’s

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INTERVIEW K A I T LY N A U R E L I A S M I T H

supposed to go back to. And then I start to get into the more granular process of going through the parts on each song and break down the parts and just practise those for a long time.”

Inner Ear On a perhaps more esoteric level, the meditative qualities of Kaitlyn Aurelia Smith’s music, particularly on The Mosaic Of Transformation, are for her directly linked to tuning into what she calls her “inner ear”. During lockdown, Smith has been doing online consultation sessions teaching others how to make music and clear their creative blockages through what is essentially a meditation process. “These songs all were made in kind of like this process of internal listening,” she says. “‘Cause I’ve always had a relationship with my inner ear and, y’know, writing music through closing my eyes and listening. But this album in particular was the most I’d used that. I would do sessions of just listening for music inside and then try and create it with the synthesizers.” That sounds quite, er, hard? “I think it’s only hard in the beginning when you can’t hear the music,” she reasons. “Honestly, I think the hardest part is in the beginning, learning what are the fundamentals that make up sound. But then once you start to practise, like, how many sounds make up the sound of a door opening, then it gets easier to break down each sound that you hear. To be like, ‘OK, it’s really six sounds layered and they’re staggered in this way.’” Which begs the question: how close does she get to realising the sounds in her head on her recordings? “It takes me a long time,” she laughs. “And I rewrote this new album, like, 12 different times because it just wasn’t there yet. But what I’ve shared now was what I heard internally. And so, I just keep going until it’s there, and take a lot of breaks.”

Home Studio In her studio at home, Kaitlyn Aurelia Smith, given her complex system of synths, uses various interfaces in conjunction with Ableton. “I have four different interfaces that I use,” she explains. “I use the [Universal Audio] Apollo, I use a Native Instruments one, and I use two different types of MOTU interfaces. Just for different sections in the studio. I use the Apollo for the UAD plug-ins. Then the Native Instruments

36

Despite being a Buchla devotee, other instruments make regular appearances in Smith’s work. Pictured are her Moog Grandmother, EDP Wasp, Roland SH-101 and, below, a Sequential Circuits Prophet-5.

one is really nice for performing live. The MOTU ones, you can send CV out. So, they all have just different functions.” Monitors-wise, Smith has Genelec M040s and a 7360A subwoofer. “Yeah, I really like those a lot. They’re great. I haven’t changed actually since I got those. They’ve just been really seamless in the mixing process. When I get a song sounding really good on those and then I go and listen on headphones or in the car or on my laptop, it feels like a pretty seamless transition.” While many of Smith’s tracks are instrumental, a handful feature her vocals. On The Mosaic Of Transformation, one song in particular, ‘The Steady Heart’, stands out with its choral layering of her voice. Smith name-checks her Peluso microphone specifically, but as with the rest of her collection, she isn’t particularly interested in mic manufacturers or model

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numbers. “I have eight different vocal mics and I just have my own names for them,” she laughs. “I don’t have a set vocal chain. It depends on the song. I used all eight on ‘The Steady Heart’ and I heard that way of layering in my ears and I just sang it. There’s no processing. It’s just dry voice.” In contrast, live, particularly going back two or three years, Smith’s vocals were much more processed than they are on her current album. Initially, she struggled to find a harmoniser that could create the modal harmonies she was after. Her solution was to create 27 microtonal pitch-shifted tracks within Ableton and use a Novation Launchpad to control her vocal input by playing chord-like shapes to manipulate the harmonies. “It’s just me in real time changing the pitch of my voice,” she says. “I like using the Novation Launchpad and how you can map it to whatever you want. I made all those

ones and Waves and UAD plug-ins. “I really don’t have a set system. It’s a process of I think about what I’m gonna do and if I need an EQ, then I try lots of different EQs until I find the right one. I think about, ‘Which kind of EQ do I need? Do I need a parametric EQ?’ And then I do research on what are all the possible ones out there. So, sometimes it’s just the Ableton ones, sometimes it’s the UAD ones, sometimes it’s Waves. It really depends.” Speaking of plug-ins, has Smith tried Arturia’s software version of the Buchla Easel? “Yeah, I have tried it,” she says. “To me, none of the emulators sound like it. They have qualities that I can see are inspired by it. But it definitely doesn’t sound like it to me.”

Collaboration

microtonal harmonies that I would play as I was singing and then I go through the Music Easel.” Of course, having only one Music Easel at her disposal when performing live, Smith compensated for the layering of the Buchla that features on her albums by sampling its output live into a Teenage Engineering OP-1. “Most of the time, whenever I’m using anything live, it’s as an extension to the Buchla,” she stresses. “So, I’m just finding ways to create more versions of the Buchla. “And then also for live,” she adds, “because you can create in Ableton your own version of unity [gain-staging] on each fader, you can fully mix how you want all of your tracks to be. I tend to have a lot of dry tracks and then I’ll have a lot of record-enabled tracks for the Buchla that have different processors on them. Then I set how I want [the processors] to be mixed and I make each of those faders the new version of unity, so then when

I map it to my hardware faders, it’s at the spot it’s supposed to be mixed at.”

Mixing Back in the studio, when it comes to mixing, Smith chooses to zone in fully on one sound element at a time before stepping back to look at the bigger picture. “I mostly do mixing as I’m going,” she says. “To me, mixing is a very important part of the composition and each sound has its own way of being mixed that communicates something. So, I’ll fully mix one sound and then work on the next one. “I’m really sensitive to the way that other people mix sounds, because it’s such an important part of the composition. Any time when I’ve had someone else mix, it changes the communication of the song. So, I tend to be the one who mixes it.” When it comes to mix EQ plug-ins, Smith moves between the stock Ableton

In 2016, Kaitlyn Aurelia Smith collaborated with ground-breaking Buchla artist Suzanne Ciani — who has been making records since 1970 — on the album Sunergy, firmly underlining her place in the lineage of exploratory modular synthesists. “That was an enjoyable process and she’s such a wonderful person,” says Smith. “It was a really neat experience to collaborate with another Buchla synthesist, cause that was the first time I had tried working with another Buchla player. I think I learned just how fun it is to play two Buchlas at once.” Looking to the future, are there any areas in particular that Smith would like to move into sonically? “That’s still being born for me,” she says, “‘cause I feel like I’m still in the process of sharing this album. And because I haven’t really shared this album live yet, that space hasn’t really opened up for me. “Usually in the beginning of a project, I start to collect adjectives and textures for the next one. And so, I’m in that process right now where I’m starting to get textures. But they’re not in a language form yet. When I hear a sound in the world, it’ll pique my interest, but I won’t know why yet.” So, finally then, any specific tech purchases that she still wants to make? “Not at this time in the current economic state,” she laughs. In other words, like many of the rest of us, she is working with what she’s got. Given the amount of gear she has at her disposal, though, we can expect from Kaitlyn Aurelia Smith many intriguing sonic adventures to come.  

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ON TEST

PreSonus Quantum 2626 Thunderbolt 3 Audio Interface The newest addition to the Quantum range offers a plethora of I/O and excellent latency performance for a very reasonable price. CHRIS KORFF

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hen it comes to audio interfaces, PreSonus have a long history of adopting new protocols. They were among the first few companies to make a Firewire interface (the 2003 FireStation, which was also the first to make use of Yamaha’s mLAN protocol), and more recently they’ve been keen early proponents of both USB 3.0 and Thunderbolt. Their first interface to use the latter was the PreSonus Quantum, which took full advantage of Thunderbolt’s data transfer speed to provide remarkably low latency. You can read our glowing review of it in the September 2017 issue. The Quantum is still available, but is now part of a range of Quanta that includes the Quantum 4848 (a no-frills but high-I/O-count

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device designed to work with analogue consoles) and the subject of this review, the Quantum 2626.

Quantum Mechanics One thing that sets the Quanta apart in a very crowded market is that they all dispense with DSP. PreSonus reasoned that if they can get latency low enough, there’s no need to include a DSP mixer in the interface itself, since the DAWs we all use have far more powerful mixers of their own. It’s an attractive prospect: rather than plugging into the interface, routing the physical input to a DSP mixer channel, piping that into your DAW while also sending it back to a DSP foldback bus, and routing your DAW back into the DSP mixer, via the foldback bus and thence to a physical output — all so you can hear yourself and your backing tracks while

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you record — you just patch the Quantum straight into your DAW channels and do all your routing and monitor mixing in there. No more Alt+tabbing to tweak your monitor mix, or loading up DSP mixer scenes that may or may not work with your

PreSonus Quantum 2626 €599 pros • Excellent latency performance. • Well built. • Plentiful I/O. • DC-coupled outputs. • Insert points for channels 1 and 2. • Great price for what’s on offer.

cons • Headphone output doesn’t quite go fully off. • No Thunderbolt cable included.

summary The PreSonus Quantum 2626 offers excellent latency performance and a generous amount of I/O for a very reasonable price.

Bundled Software The Quantum 2626 ships with a licence for Studio One Artist, as well as the 2020 Studio Magic bundle, which includes Lite versions of Ableton Live and Arturia Analog Lab, plus a good selection of plug-ins from Brainworx, iZotope, Cherry Audio, Lexicon, Mäag, Native Instruments, SPL and more.

current DAW project. Routing, headphone mixes, monitor outputs and levels: all are recalled perfectly the moment you open a DAW project. The remarkable latency performance of the original Quantum made the above a practical reality but, at its £1000-plus price tag, it was competing with manufacturers who’d turned DSP into an artform. Universal Audio, in particular, have spent years considerably sweetening the DSP deal with some outstanding plug-ins, and their powerful low-latency Console DSP mixer exemplifies the opposite approach. Announced earlier this year, the Quantum 2626 promised all of the blissful simplicity of the original Quantum but at just over half the price. What’s the catch?

Ins & Outs Despite not having any internal routing or signal processing, the original Quantum still boasted some neat digital extras, many of which have been lost on the 2626. So the digital control of preamp gains and assignable monitor controller are gone, the latter meaning you can’t perform surround-sound monitor control without getting a separate monitor controller. Also absent are the talkback section, the eight-segment LED metering, the individual phantom power switching, and the separate D-A converters for

the headphone amps — the two on the 2626 always follow the main outputs. The all-important I/O, however, remains intact: the Quantum 2626 features eight of PreSonus’ XMAX preamps, all of which are accessed via combi inputs on the front panel. The first two can accept high-impedance instrument signals, and also have balanced, pre-conversion insert points on the rear. Also round the back are eight line outs, with outputs 1+2 mirrored on dedicated monitor

outputs and controlled by the front-panel attenuator. In terms of digital connectivity, you’ve got stereo S/PDIF I/O, two pairs of ADAT ports, word clock I/O, a pair of proper five-pin MIDI sockets (In and Out), and one Thunderbolt 3 port (see box). The ADAT ports can either provide eight I/O at higher sample rates (ie. in S/MUX mode), or be used in independent pairs to add 16 extra I/O at base sample rates. Given the desire to bring the cost down, I think the omissions have been very well chosen. I rarely need a talkback mic or surround monitoring, and I’d venture that’s the case for most home/ project-studio owners. I have a nice big screen in front of me that can tell me more about my recording levels than an LED meter ever could, and the 2626’s two banks of phantom power switching, rather than the Quantum’s individual phantom switches, are fine by me. I also don’t mind the lack of digital control over the preamp gains: the analogue knobs on the front panel are sturdy, and work smoothly and predictably. I had no trouble matching gains for stereo miking, for example. If digital gain control is important to you,

Thunderbolt 2 Vs 3 The original Quantum was a Thunderbolt 2 device, and sported two mini-DisplayPort sockets, allowing you to daisy-chain other Thunderbolt devices (including additional Quanta) off it. The 2626 uses Thunderbolt 3 via the newer reversible USB-C-type socket, but it has only one port. You can use it as part of a multi-Quanta setup, but it must be at the end of the chain. The 2015 iMac I used for this review only has Thunderbolt 2 mini-DP ports, but the 2626 manual says you can use it on TB2, provided you use the official Apple Thunderbolt 3 (USB-C) to Thunderbolt 2 adaptor. That adaptor is bidirectional, so while Apple chiefly describe it as allowing you to plug older devices into TB3 ports, it also does the job of letting you use the 2626 on TB2-equipped Macs. I researched all of this quite carefully before buying the official

Apple adaptor because it costs nearly £50. And given that PreSonus, in common with other Thunderbolt interface manufacturers, don’t provide a Thunderbolt cable with the 2626, I was down 80-odd quid by the time I’d also bought an official Apple Thunderbolt USB-C-type cable. What the manual doesn’t make clear is whether there’s any performance or latency penalty incurred by using the TB2 adaptor, so I asked PreSonus. Happily, they told me that performance is identical whether you’re using TB2 or TB3, because their bus speeds are the same. (Thunderbolt 3 has higher data throughput, but audio isn’t particularly demanding data-wise so TB2’s bandwidth is perfectly adequate). Apparently there is a small increase in latency when using the 2626 via Thunderbolt 1, but I didn’t have a TB1-equipped computer on hand to test that.

w w w . s o u n d o n s o u n d . c o m / September 2020

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ON TEST PRESONUS QUANTUM 2626

For a ‘cut-down’ interface the Quantum 2626 enjoys a generous provision of I/O.

though, I should mention that the 2626 can be paired with a PreSonus DigiMax DP88 ADAT expander (or two), which does have software-controlled preamps.

Firing Up Setting up the Quantum 2626 was a painless affair. You need to register it with PreSonus, after which you can log in to the PreSonus website and download Universal Control, which is a simple app that takes charge of such exciting things as firmware updates, sample rate and sync options (internal, ADAT, S/PDIF or word clock). This is also where you can control your DigiMax DP88s, if you have any (if you’re a Studio One user, you can even control them from your DAW). You hook up the external PSU (which has a twist-locking connector), plug into a spare Thunderbolt port (via any necessary adaptors), and the light on the front panel changes from red to blue to show that sync has been achieved. In my case, UC then told me that the Quantum 2626 was due a firmware update, which was again a simple process, taking all of about 20 seconds. The absence of a routing matrix, DSP mixer, built-in effects and so on make using the Quantum 2626 refreshingly simple. You select your sample rate, open your DAW of choice, set your buffer size (I used the lowest option of 32 samples), and you’re set. Everything behaved exactly as expected, with the one exception being that the headphone outs never quite went fully off; even at ‘zero’ there was still a very low-level signal. In terms of sound quality, though, there

was nothing to fault. Indeed, the noise and distortion specs appear completely unchanged across the board compared with the original Quantum, and they are certainly more than good enough to not impose themselves. All types of signal I tried it with — mic, line and instrument — were handled perfectly well. I tend to really dig in when I’m playing bass, for example, and can easily overload lesser instrument inputs and DI boxes, but that wasn’t the case here. I even had to turn the gain up a bit on the 2626, which made a nice change from having to turn my bass down or find a device with a pad to put between me and the input. As important as the sound, however — and especially with an interface bereft of any internal routing — is latency. Guitarists, bassists and keyboardists can generally live with a bit of latency

set up a separate foldback mix would be the icing on the cake. Next, I repeated the latency test that Sam Inglis did in his review of the original Quantum, and came up with identical results: 0.9ms input latency and 1.0ms output latency at a 32-sample buffer size, according to both Reaper and a loop-back test. Which goes to show that, in the 2626, PreSonus really have left all the important parts of the Quantum intact.

Get DSP Hence Ever since I’ve had a home studio, I’ve flip-flopped between deciding that I do and don’t need a mixer more times than I care to count. I’ve had a number of analogue and digital consoles, and I’ve had some enormously complicated DSP soundcards (I’m looking at you, Creamware), and they have all come with their own joys and frustrations. But other than the ‘niceness’ of having real faders and knobs to tweak, the main reason I’ve kept going back to mixers is the lack of latency. About half a year ago, I decided that mixers were good again, and rejigged and rewired my ever-changing lab to accommodate one. Since using the 2626, however, I’ve actually come to enjoy using my DAW more: having only one place to record, to monitor, to add effects and to noodle around has made me more focused and, I think, more productive. For one thing, opening projects has become more of a pleasure than a chore, now that I no longer have to remember how my mixer was set up whenever I saved ‘piano & bass sketch v43’. And with studio time in ever scarcer supply, that has real value. In terms of workflow, the Quantum 2626 has been perhaps the least imposing bit of equipment to ever grace my rack, and certainly the simplest multichannel interface I’ve used. It could well be the interface that makes me flip-flop for the final time.  

“Since using the 2626, I’ve actually come to enjoy using my DAW more.”

DC Universe All Quantum-series interfaces have DC-coupled outputs, meaning that, with the right software plug-ins, you can send control voltage signals to analogue and modular synths. I didn’t get a chance to try this out myself, but there’s an excellent feature about it elsewhere in this issue.

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quite happily because they’re used to being a few feet away from their amps (each foot adding a millisecond or so of delay), and for virtual instruments the round-trip latency is pretty much halved because there’s no input latency to deal with. Vocals, on the other hand, are more of a challenge. We’re so used to hearing our voices through our own skulls that even a handful of milliseconds can be remarkably distracting when you’re trying to sing. So for my first test of the 2626’s latency I got a small analogue mixer, plugged a mic into it, sent it to the Quantum 2626, and routed the direct analogue signal to one bus and the interface return to the other. I sang into the mic, alternately PFL’ing the direct signal and the DAW return busses. And once I’d matched the levels, I was hard‑pressed to tell the difference. I make no claims to be a good singer, but if I ever felt compelled to commit my crooning to WAV, I’d have no qualms at all about monitoring from my DAW. And the enormous convenience of not having to

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££ €599 including VAT. WW www.presonus.com

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ON TEST

VSL Big Bang Orchestra Sample Libraries

The Viennese maestros go intergalactic with their full-scale orchestra range. VSL Big Bang Orchestra pros • The first four titles feature a 70-piece orchestra recorded with all musicians playing together. • Dorado and Eridanus put the ‘bang’ in ‘big bang’, while Fornax offers unique tuned percussion colours. • Ganymede adds a flexible SATB choir of 48 singers. • The libraries collectively cover a comprehensive range of articulations, textures, effects and phrases. • Everything was recorded from multiple mic positions in a state-of-the-art soundstage.

cons • Fails to answer the question, ‘What was there before the Big Bang?’

summary Vienna Big Bang Orchestra pushes the boat out with a massive 70-piece orchestra, a choir and six percussionists letting rip in a series of themed libraries. Recorded in a historic soundstage from 10 mic positions, this extensive range of ready-to-play, pre-orchestrated performances and textures takes the strain out of creating dramatic scores. Practical and affordable for beginners, BBO also has plenty to offer pro users.

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DAVE STEWART

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aking advantage of the large scoring stage which lies at the heart of their creative operation, Vienna Symphonic Library have pulled out all the stops for their latest project. The Big Bang Orchestra series features a massive 70-piece orchestra, six percussionists and 48 singers performing effects, textures, hits, chords, clusters, arpeggios, runs, riffs, rhythmic patterns and single-note multisamples, all recorded from multiple mic positions in VSL’s historic Synchron Stage. Designed for maximum impact with minimal effort, these pre-orchestrated starter editions are available at entry-level prices, an incentive for new users to embark on a personal exploration of the Vienna musical universe. It’s one thing to record an orchestra in separate sections, quite another to capture the whole shooting match in one pass. With the latter approach you experience the sonic shock wave of a large musical ensemble letting rip in the same space, which can rival the noise of a jet engine — strings, brass

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and woodwinds interacting in a complex, intense and sometimes overpowering live mix. This uproarious racket can be heard in full effect in BBO: hence the series’ name, and its marketing slogan, ‘Have a Blast!’ The libraries’ musical content was composed and orchestrated by Johannes Vogel and performed by the Synchron Stage Orchestra, an elite squad of instrumentalists and singers hand picked from world-class Viennese orchestras. At the time of writing the series comprises eight themed collections, with more to come: all run exclusively on VSL’s free Synchron Player, but are not compatible with the company’s older Vienna Instruments player. In this review we’ll take a look at each BBO library with a view to giving an overview of the series.

BBO Free Basics BBO Free Basics (1.5GB) contains classic articulations played live by all 70 orchestral players. Instruments are blended across a playing range of A#1-D6, but since all notes incorporate multiple octaves you actually hear the orchestra’s full seven-octave span, from

Capricorn provides ready-to-play, energetic orchestral phrases played at three tempos. A built-in time-stretching feature lets you change the tempo to fit your arrangement.

a thunderous low A#0 up to a tinnitus-inducing D7. The low end is positively stentorian: on hearing the rasping bass trombones and low strings rattling my music room windows, my partner remarked, “It sounds like ready-made film music — really big, as if something amazingly momentous is about to happen.” I guess that means these samples tick the requisite ‘cinematic’ box. A smallish menu of presets includes three dynamic looped sustains and dramatic short-note stabs, which double as an optional marcato attack layer. The long notes are surprisingly versatile: a great timbre for loud, majestic fanfares, they also sound beautifully solemn and sonorous when played quietly and sparsely in the mid-range. In addition, there are tremendous swells and fast chromatic octave runs to a short accented target note, performed in six keys. All good riotous fun, delivered with gusto and attitude. This library would be a good first step for anyone wanting to start a collection of full-orchestra performance samples. Though available as a free download, a ViennaKey USB protection device (which can be used to store all your VSL licenses) is also required.

BBO Andromeda Essentials Named after the nearest major galaxy to the Milky Way, Andromeda (11GB) expands the menu of essential styles and adds close mic positions. New articulations include climactic crescendo swells (played longer than those in Free Basics), trills (which sound like a swarm of gigantic locusts when played by a whole orchestra), and earth-shaking massed tremolos. In a similar agitated vein are fast note repetitions played in a choice of three tempos, which generate a sense of

hectic rhythmic excitement despite their lack of a pronounced pulse. If you need a scary motif of the ‘shark approaching!’ variety, there are some terrific semitone falls — classic Jaws fodder when played in the bass, and decidedly psychotic at the high end. For less troubling scenarios, Andromeda offers its own freshly recorded set of long and short notes, very nicely played and well tuned. The shorts work well in rhythm passages, while the stately sustains combine instant gravitas with some volcanic tutti bass notes. This library provides optional marcato attacks and a separate piccolo layer for all artics. I’m glad the piccolo is an optional extra as its piercing, ultra-high-pitched shriek gets tiresome quite quickly! Long-time VSL users will be pleased to see the company’s trademark sforzato and sforzatissimo attacks included in the artics menu, along with expressive ‘soft low brass’ and ‘soft swell’ performances. All styles are performed by the entire orchestra and available within a single preset via an elaborate keyswitch system.

BBO Black Eye Phrases & FX Were it not for the series’ galactic theme, one might assume Black Eye owed its name to the enormous punch of some of its contents, in particular its grandiose orchestral hits. These offer some unusual

variations on a familiar theme: built-in octave shifts spread the impact of the ‘flam hits’, while the ‘delayed’ style has the lower register parts sounding slightly later than the high instruments. But if you simply want the no-nonsense wallop of a straight orchestra hit, the ‘graced’ version provided here does the trick. The phrase samples include a nicely voiced set of staccato chord stabs and some fantastic tutti octave runs. Brilliantly played in a choice of major and melodic minor scales, the runs are performed across the entire A#1-D5 range, so in most cases the same scale can be played in four different octaves. ‘Rips’, a quick triplet run up or down to a sustained target note, are also excellent, and have both mystery and comedy potential. Standout effects phrases include the massive sforzando attack of ‘echoes’ and the chaotic swirling of the octave trills, the latter exciting in the high register. The slow ‘microtuning’ patch also sounds fantastically menacing in the low end, especially when smeared over a fistful of adjacent semitones. ‘Seagull arpeggios’ (a valiant attempt to double spooky harmonic-series string arpeggios with wind instruments) are arguably less successful, but I daresay imaginative composers will find a use for them. As ever, atonal clusters are a great horror-film sonority. I liked the ‘cluster to

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ON TEST VSL BIG BANG ORCHESTRA

The BBO Free Basics library features the 70-piece Synchron Stage Orchestra playing a set of essential articulations.

root’ artic, in which a discordant pile of pitches collapses satisfyingly into a single note sustain, and admired the marcato ‘Clusters B’ sample mapped on D#5. This blaring, screaming voicing would be a fitting end to the bonkers symphonies you’ve all no doubt been cooking up in the lockdown period. All in all, this substantial 11GB collection is a great source of inspirational orchestral textures.

BBO Capricorn Symphonic Riffs By far the largest BBO library, Capricorn (42.6GB) brings you ready-to-play four-bar orchestral phrases, enabling users to unleash full-scale symphonic riffs with one key press. A grand total of 24 phrases are divided into main, low and high patches, selectable on the fly via keyswitches. Played in a choice of three tempos at two dynamics and chromatically mapped for use in all keys, these looped ostinato

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patterns repeat seamlessly for as long as you hold down a key, and sync accurately to a click. The riffs centre on the key note and often feature simple minor-key phrases. Highlights include ‘Main riff 1’ (pumping, repeated single eighth notes played with a 3+2+3 accent pattern) and ‘Low riff 1’, the same idea played as 3+3+2. ‘Main riff 5’ is another powerful eighth-note ostinato with a vaguely Arabic flavour, the galloping ‘Main riff 6’ has a historical-drama vibe (think jousting

matches, Battle of Agincourt and Robin Hood: Men In Tights), while ‘High riff 3’ is a stirring, triumphal fanfare-like triplet motif for all occasions. A comprehensive keyswitch system allows you to select tempo, release type (one of which adds an automatic final note to the riff), and eight variants each of the main, low and high riffs. The uppermost C6-C7 zone plays a combination of the selected low and high riffs, either of which can be muted with a ‘tacet’ keyswitch.

Vienna Synchron Player Designed to handle surround-sound formats up to 9.1, VSL’s Synchron Player takes over where the stereo Vienna Instruments player software leaves off. The latter’s matrix switching system has been replaced by the ‘Dimension Tree’, a series of switches arranged in columns. Each column contains a number of colour-coded slots displaying the name of a musical ‘dimension’ (ie. playing style or patch name): the same colours are shown on the interface’s keyboard, making it easy to learn keyswitch layouts.

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Patches can be layered or crossfaded by clicking on the ‘stack’ icon at the top right of the dimension column. Other facilities include a resizable screen, a great collection of built-in effects (including an algorithmic reverb), and a non-destructive time-stretching feature, which lets you automatically sync riffs to your host tempo. In other good news, the software now works standalone on your desktop as well as a plug-in in your DAW. What’s not to like?

ON TEST VSL BIG BANG ORCHESTRA

The Vienna Synchron Player’s ‘Dimension Tree’ is a flexible switching system which shows all the available playing styles for an instrument. Coloured keys on the GUI keyboard correspond to the colour-coded ‘dimension’ slots. Note that the currently selected keyswitches (in this case, A0 and C1) are displayed with a darker colour.

These energetic patterns have great propulsive feel and a distinct rock sensibility. Most would make good action and adventure cues, while some have a dramatic classical vibe. I was pleased to see a full score of the contents provided in the user account ‘Tutorials’ section, which is a great educational aid for students of orchestration.

BBO Dorado Percussion Ensembles Composers looking for the ultimate ‘big bang’ will likely head for BBO’s percussion section. To that end, Dorado (13.7GB) provides a six-player percussion ensemble performing essential articulations. The

Alternatives Of all the mega-sized orchestral collections out there, I’ve yet to find one (present company excepted) that provides a full symphonic contingent of strings, brass and woodwinds playing together live. That makes the Big Bang Orchestra’s Free Basics, Andromeda, Black Eye and Capricorn collections unique, and given the scale of the project, it’s remarkable that VSL are giving away the first title and selling the others at affordable prices. Concentrating on titles in the lower price range, an alternative to BBO Dorado would be Red Room Audio’s Saga Acoustic Trailer Percussion, recorded from two mic positions. For standalone orchestral percussion patterns, Cinesamples Deep Percussion Beds 1 has a similar stylistic slant to BBO Eridanus and offers more phrases, but only one stereo miking. No other sound library matches the pianos-plus-perc instrumentation of BBO Fornax, but affordable alternatives to the BBO Ganymede choir exist in the shape of Soniccouture’s All Saints Choir and Soundiron’s Requiem Light.

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group was recorded live with pairs of players in spaced left, centre and right positions, creating a panoramic soundscape; all performances were played at five dynamics with hard and soft beaters in a generous range of techniques displayed in an on-screen ‘key info’ chart. A mouth-watering collection of instruments was trotted out for this project: the battling ‘Monster Drums’ combines two large taikos, two orchestral bass drums and two surdos, ‘Thunder Toms’ raises the roof with six differently sized concert toms, while three sets of piatti cymbals make an enormous splash in the ‘Super Crashes’ patch. Add a crisp snare ensemble, suspended cymbal crescendo rolls and the ominous, hellish clang of tam tam gongs, plate bells and anvil, and you have all the ingredients for the big, combative ‘cinematic’ percussion sound beloved of today’s media composers. It’s not all epic: Indian mridangam, log drum and temple block create a more intimate world-music atmosphere in ‘Special Drums’, and ‘Small Metal’ features a child-friendly mix of triangles, jingle rings and cymbals. Quiet hits reveal some wonderful subtle, resonant sonorities which would shine in more delicate musical settings. The library also features some transformative processed mixes: ‘voltage shot’ brings to mind the classic 1980s gated snare effect, ‘low boom’ and ‘low explosions’ are ideal low-frequency signals for the surround ‘sub’ channel, and when applied to crash cymbals, ‘phaser shot’ sounds like the opening of a psychedelic concept album. Imbued with a classy hall acoustic, these big, resonant hits sound like orchestral percussion should, a welcome alternative to the controlled studio sound of VSL’s earlier libraries.

BBO Eridanus Percussion Riffs If you fancy adding percussion parts to Capricorn’s symphonic riffs, Eridanus provides a ready-made solution. Composed by the versatile Johannes Vogel, these driving patterns exactly match the rhythms of the Capricorn set, and can also be used alone to energise any arrangement that requires propulsive orchestral percussion. Featuring the combined force of up to six percussionists playing together, the 6.4GB phrase library replicates the playing positions and tempos used in Dorado. Main and low riffs are played on

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The Synchron Player’s large suite of effects includes a convolution reverb based on VSL’s MIR Pro technology, delay, rotary speaker, distortion, low- and high-pass filters and a powerful compressor. Shown here are the multiband EQ section (top) and the mixer ‘power pan’ control, which allows you to adjust stereo width as well as L-R balance.

large taikos, bass drum and concert toms, while the high riffs (played in a choice of sticks and brushes) feature an unusual, exotic-sounding blend of small tom, cymbals, snare drum, congas and bongos. Single hits are also provided so you can add an emphatic ‘bonk’ to a pattern at any point! Layering ‘Main riff 3’ with the corresponding Capricorn phrase produces instant film music: a triplet-based orchestral figure reminiscent of Ravel’s ‘Bolero’, underpinned by big, booming drums played with a 12/8 feel (think John Williams’ Superman theme). Contrasting with the low drums’ warlike thunder is the less identifiable hand-drum-like timbre of the high riffs. Played with impressive precision, their tight, urgent performances include some great brush patterns.

BBO Fornax Pitched Percussion Burning bright in BBO’s night sky, Fornax contains a unique set of sounds created by blending tuned percussion with three grand pianos. These instruments (a Steinway D-274, Bösendorfer Imperial and Yamaha CFX) are not the sort you might find in the back room of a pub:

BBO’s 70 musicians perform live in the historic Vienna Synchron Stage.

they’re collectively worth more than most people’s houses, so getting all three for less than a hundred quid could be considered quite a bargain! The ‘Drums and Pianos’ preset pits these expensive concert grands against a large taiko drum, timpani and a 36-inch bass drum to deliver thumping, dimly clangourous straight notes, dramatic crescendo rolls and some devastating, enormous-sounding cluster hits. For more subtle settings, the ‘Big Timpani’ soft beater performances add a gentle percussive kick, while the bass notes of ‘Low Bells and Pianos’ (which mixes the chimes of tubular and plate bells with the pianos’ grumbling low octaves) would work a treat in a film music cue. Though its tuned gongs hail from Burma, ‘High Gongs’ evokes the jingling timbre of Balinese gamelan by adding triangles and a suspended cymbal. I found that muting the cymbal’s spot mics creates a lighter, more delicate and pretty sound. ‘High Bells’ is also very pleasant, an ear-catching chime which adds Eastern spice to the formal tones of European tuned percussion. You could use this patch to trace an orchestral top line, or even add to a pop chorus. While this 17.4GB library’s distinctive character stems largely from its high‑range instruments, the heavier drums and timpani presets should appeal to those operating within ‘cinematic’ norms. For that style, the ‘Spooky Melodies’ and ‘Doomsday’ mixer presets add explosive aggression to the large

drums, while some lovely ethereal effects are also available if you want to sprinkle fairy dust on the bell chimes.

BBO Ganymede Choirs The first leg of VSL’s alphabetical stroll through the cosmos fetches up on Jupiter’s largest moon. The first BBO library to contain true legato recordings (the brilliant interval-specific technique invented by VSL back in the day, now an industry standard), Ganymede features a mixed-voice choir of 48 singers divided equally into sopranos, altos, tenors and basses. The four sections were recorded individually and also together in real-time ‘tutti’ performances. I had a great time playing the ‘stacked’ patches, which seamlessly map the sections over a full C2-D6 range suitable for two-handed playing. The looped three-dynamic sustains are blessed with immaculate tuning and sung in a plain style free of operatic ‘wobble’, so you can use them for any kind of music without fear of culture clashes. I also liked the graceful dynamic rise and fall of the ‘espressivo’ performances; by contrast, the loud, emphatic attack of the sforzato articulation is ideally suited to epic, big-screen productions. This being an entry-level title, there’s no ‘word-building’ feature. Latin phrases are also mercifully absent (I think if I hear another sampled choir piously intoning ‘Sanctus’, I’ll go crazy — wake me up when they start including rude football chants). On the vowel front,

these guys simply sing ‘ah’, a lyric few could take issue with. The tutti effects contain some great material: dense, atonal clusters corresponding to those in Black Eye, cluster chords which slowly build up from low to high voices and vice versa, histrionic glissando slides and dynamic ‘hah’ and ‘hooh’ shouts performed from a sinister whisper to an exhilarating roar. This excellent 7.8GB library also includes sustained tutti major and minor chords in a full two-octave voicing, creating a triumphal and majestic effect.

Space Race As I write, news comes down the wire that Sir Richard Branson’s Virgin Orbit rocket has failed on its debut flight. Ho hum. The report adds, “The history of rocketry shows that maiden outings very often encounter technical problems.” Over in mainland Europe, some space explorers are enjoying better fortune. As SOS’s newly appointed rocket correspondent, I’m pleased to report that VSL’s debut venture into the far reaches of the musical cosmos has so far succeeded admirably, and looks set to give its buyers an exciting and fruitful voyage. Commencing countdown... ££ BBO Andromeda €130, BBO Black Eye €110, BBO Capricorn €95, BBO Dorado €95, BBO Eridanus €75, BBO Fornax €95, BBO Ganymede €160. Prices include VAT. WW www.vsl.co.at

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TECHNIQUE

Setting Up & Using Jamulus The free and open-source Jamulus app lets you play along with other musicians over the Internet in real time. We guide you through it. CHRIS TIMSON

L

atency. Anyone concerned with trying to create live music over the internet will have become very familiar with that word. As I explain in my article ‘How to Make Zoom Work for Music’ (SOS June 2020), latency is like death and taxes; you can minimise it but you can’t avoid it completely and normally it defeats attempts to play live together. As anyone who has tried it with the various video conferencing applications like Zoom and Teams will know it results in everyone being out of time with everyone else, with consequent cacophony! So you may be as surprised as I was to learn there are a small number of applications where the developers refuse to admit defeat on this and are determined to allow musicians to play together live across the net. Among these are JamKazam, Soundjack and the subject of this article, Jamulus.

Environment The first thing to note about these applications is that they all demand a lot of your environment in order to ensure the only source of latency they have to deal with is the Internet itself. You have to watch out for the following factors: Hardware: A powerful computer is always nice to have, but these apps don’t put too much demand on the host machine. Any

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decent computer from the last few years should do fine. For Jamulus, Windows, Mac and Linux are all good. Tablets and phones, unfortunately, are right out! A separate audio interface is a necessity in order to provide a low‑latency connection to the computer. In our own usage, UAD’s Apollo hardware worked well, while a Blue Icicle XLR-USB adaptor wasn’t so effective, adding pops and clicks. For Windows, Jamulus recommend an audio interface with a native ASIO driver. Decent microphones to plug into the interface give the best sound, obviously, and are well worth having.

Networking: You need a fast Internet connection. Jamulus specify a minimum of 1Mbps both up and downstream, but faster is better and a low ping time is essential. Jamulus recommend the ping time to the Jamulus server (see below) should be no more than 40ms, and the lower the (very much!) better. All of these applications make the same recommendation with regard to Wi-Fi, and that is simply not to use it. You must set up a wired Ethernet connection between your router and computer. They don’t mention Homeplug and other powerline solutions but I would be inclined to avoid them too if you possibly can.

Screen 1: What you’ll see when you first open the Jamulus client.

Distance: That is, geographical distance between the participants. More on this below, but it is very important to be as close as possible, geographically speaking. Headphones: You need them. Jamulus doesn’t screen out sound from your mic the way that, say, Zoom does, so your speakers must be off or otherwise you’re heading for feedback big time. Much of this would simply not be available to the average person, but is likely meat-and-drink to the readers of this esteemed magazine.

Free & Easy

Screen 2: The settings page, where you tell Jamulus which audio inputs to use.

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And so to the application itself. Jamulus is an open‑source product developed under the GNU General Public Licence and, as befits

Screen 3: After you hit the Connect button, you’ll see a list of public servers in order of ping time.

such software, you don’t need to create an account with anybody to use it! The way they achieve this is ingenious. The application has two parts: a server and a client. In order to work, a client has to attach to a server. It sends sound to the server and plays the sound it receives back from it. A server is a very simple beast in that any sound received by it from any attached client is bounced out to all attached clients. Of course there’s a little bit more to it than that and we’ll talk about mixing the server performs later, but you’ll see immediately that with a public server the idea of a private session is not possible. So connecting to a public server is fine if you’re looking for a jam session to join in with and you like the idea of jamming with strangers, and indeed a lot of people use Jamulus for exactly that. But if what you want is to have a quiet little rehearsal with the rest of your band then it’s not going to cut the mustard. The solution Jamulus have come up with is quite elegant and is known as the ‘private server’. How this is achieved is another thing we’ll be looking at later.

MicPre

Opto-Compressor

Equaliser

Jamulus Client First we’ll talk about the client and how you install and set it up. An installer can be downloaded from the Jamulus website. On Windows it will install both Jamulus client and Jamulus server; on the Mac you’re offered a choice. You will always need the client but the server is only for if you are intending to run one. After installation, start the client and you’ll see Screen 1. This is pretty spartan, so you need to click on the Settings button on the lower left‑hand side. This then gives you the Settings screen where lots of interesting stuff lives (Screen 2). You can see I’ve set this up to use with my UAD Apollo. This is on a Mac, but if you’re setting up on a Windows machine then you’ll need to specify the ASIO driver — as mentioned earlier, a native ASIO driver is preferable. Jamulus permits two mic inputs per client, but as I’m using

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TECHNIQUE JAMULUS

Screen 4: Upon joining a server, the mixer controls will become active.

only one mic I’ve set both channels to it. I set the outputs appropriately, and that’s just about all the setup I did. All the other settings I left at default. There’s a Profile page tucked away in the menus that you can use to provide a name for yourself and add the instrument(s) you play, which is useful for ID purposes, but by no means mandatory. Incidentally, Jamulus requires that you turn direct monitoring off for the live mics in your audio interface. To make Jamulus actually do anything you must connect to a server, so click on the Connect button and you’ll see something like Screen 3. This is a list of public servers, organised by latency with respect to you. You can see that at the time I was connecting (a Sunday morning) there weren’t too many musicians around but a couple of servers had people waiting for others to play with. Once you’ve connected to a server, the appearance of the client suddenly gets richer (Screen 4). There’s a fader for each participant, including you, so that you can mix a balance for yourself to hear in your headphones. Note that the mix is actually performed by the server, which sends the resulting stereo mix down the line to you. Were the mix to be made at the client end then each client stream would need to be sent to you, requiring lots more data and

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potential for the streams to get out of step. Now you can start playing!

Pass The Port So you’ve had a good jam session but now you want a rehearsal with your band. Obviously you can’t use a public server for this because anybody could gate crash your rehearsal. The answer is for one member of the band to install a private server. Running a private server is trivially easy, you just click on it and away it goes... Except for one thing: if you’ve started the server running on a computer

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on your own network then it will be inside your router’s firewall. This means anybody outside your firewall (like your bandmates) won’t be able to access the server. You need to open a port in your firewall; that’s not nearly as bad as it sounds, but it does need careful explaining. To most people the idea of opening a port in the firewall feels tantamount to opening the front door of your house and posting a sign saying ‘Come in and help yourself!’ It’s nothing like as bad as that. An open port is not a risk as such, indeed there are quite a few ports open already in every firewall. There have to be, otherwise the Internet simply would not work. Port 25 is open for email transmission, for instance, and port 80 for Hypertext Transfer Protocol, which is what makes the World Wide Web work. The Bad Guys are interested in open ports because whatever process of yours is running on the inside of the port receiving data through it (‘listening’, as they say) may have bugs or otherwise be vulnerable to attack. If there is no process running on the inside to receive data, the operating system simply throws away any incoming packets (including any sent by a hacker). That is, an open port is not like an open window. You can’t just peer through. I only run the Jamulus server as a foreground application, and close it down when playing has finished, so it’s not listening for very long and nothing else uses that port. As a private server, Jamulus listens on UDP port 21224. On your router you need to open that port and set it to forward to the IP address of your computer on your own network. If that changes from time to time (such as if you’re using DHCP), just set it to forward to all. Each router is different in how it lets you do this. Screen 5 is taken from the Jamulus website, and it shows the task being performed on a Linksys router, where it’s called Port Forwarding.

Screen 5: In order to set up a private server, you’ll need to open up port 22124 in your router.

Once you’ve got your private server running then you need to inform your bandmates of your public IP address on the Internet. (If you need to find this out then use the website http://ip4.me. It’s quick, free and ad-free). When you and your bandmates start their Jamulus clients and click on Connect, you all ignore the list of public servers and, in the box down the bottom labelled Server Name/ Address, you type that public IP address and click on Connect. Everything else being equal you will hear your bandmates in your headphones and you can get on with your rehearsal.

Geography In our own band rehearsals, two of us are in the same building and the third is just 10 miles away. We do have a little latency (just enough to notice, like a very, very short delay and the odd pop or click), but in the main we are able to practice our songs quite successfully. To get some idea of the impact of distance on latency I arranged a Jamulus session with friend and studio owner Bob Bickerton

in New Zealand (10am here in the UK, 9pm there). Bob had no problems setting up the Jamulus client or connecting to my private server. However, his latency was of the order of 300-400 ms. I don’t know if you’ve ever tried talking with your voice sounding in your headphones delayed by one-third of a second, but it’s next to impossible. So then we tried connecting to a public server that was geographically almost exactly midway between us (in Japan). Here we both had latency of getting on for 200ms. Conversation was just possible but playing music wouldn’t have been, so we didn’t try. One slightly strange effect we noticed was that any changes we made to the mix were also delayed by latency, so that on the high‑latency servers, changes were delayed in a quite disorientating manner until we realised what was going on. This is due to the fact that the Personal Mix faders affect the mix at the server rather than at the client end. Interestingly, a couple of times while we were connected, other people connected and then disconnected in a few

seconds. Clearly people connect to public servers, and then if what they hear isn’t what they’re looking for they drop out quickly. This sounds like good etiquette. So playing sessions with friends on the other side of the planet isn’t quite here — yet.

Conclusion Jamulus strikes me as a well-thought‑out tool for making music across the Internet, but the great beast latency is not totally defeated, and there is a definite element of luck in whether it works out for you. Having said that, if you can put your hands on the hardware necessary then do give it a go. I have not covered all the settings available for tweaking (see the Jamulus documentation for more information on that). It’s hedged around with significant requirements and geographical restrictions, but with a little bit of luck and a following wind, and provided you stay fairly local, you really can play music in real time with other musicians across the Internet.

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w w w . s o u n d o n s o u n d . c o m / September 2020

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ON TEST

PreSonus Studio One 5

Digital Audio Workstation

Studio One continues to battle the big hitters with a broad range of enhancements and new features. ROBIN VINCENT

I

t’s been 10 years since Studio One first appeared as a spunky little DAW with ambitious hopes of poaching users from the long-toothed platforms of Cubase, Logic and Pro Tools. Where some

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DAWs have a definitive focus, Studio One would like to be all things to all people, and so with version 5 it has something for everyone. We have mixer scenes and a listening bus in the console for the studio people, we have score writing and MPE editing for the composers, we

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have a whole new Show Page for live performers and a new way of paying for it in the PreSonus Sphere.

Look & Feel When you open up this fresh new version you’re greeted with the comfortingly familiar sight of exactly the same interface. There’s a tiny bit of rejigging in the Inspector channel strip and the addition of the Show button, but otherwise there are no discernible changes. I guess they felt they nailed it

PreSonus Studio One 5 £344

pros • Mixer scenes will change your workflow. • Aux Channels. • Listen bus. • Love the new-look Analog Delay. • Decent MPE Editing. • Score view. • Show Page is fabulous. • Sphere all-in bundle is great value.

cons • No printing in Score view. • No audio capture in Show Page. • No new instruments, sounds, content or plug-ins. • No included MPE‑compatible instruments.

summary Version 5 is a meaty update that broadens the appeal, improves the workflow and takes it forward. You have whole new ways to play, produce, edit and perform your music and it shows how the term DAW is becoming increasingly inadequate.

The main GUI is unchanged but some of the plug-ins have had a facelift.

with version 4 so why waste development time trying to improve upon perfection? Moving along then.

Arrange View There’s been a very welcome change to how the gain of clips is handled. With version 4 each clip had a gain handle that you could pull up and down and fade handles at the top corners for creating a basic volume envelope. With version 5 you can go to town with creating what PreSonus call Clip Gain Envelopes, using any of the transform tools from lines to sine waves to freehand. The waveform display follows along nicely, giving you

a graphical view of any changes. It’s perfect for taking out glitches, smoothing peaks or evening things out without having to get into automating the track volume and there’s plenty of room for exploring avenues of amplitude modulation. Staying with clips for the moment, PreSonus have added an option to the Advanced Options menu that says ‘No overlap when editing events’. This removes the strange greying of clips that happens if you place a clip over the top of another clip where the previous clip still exists underneath. Instead, with this option ticked the material behind the clip is deleted and the overlapping clip takes its place. I find this a lot less confusing.

The Marker Track gets its own Inspector panel with a simple list of the markers which you can jump to or rename. You can also right-click in the Inspector space and select ‘Create Arranger Sections From Markers’ to push the markers into the Arranger Track, which is an obvious shortcut now that you think about it. We also get a new Timestretch option. After years of working to perfect the ability to separate time and pitch PreSonus brings back the old-fashion way of changing the speed of playback with the ‘Tape — Resampler’ mode. With this option selected on a track any tempo changes will be reflected in the slowing down or speeding up of the audio in a tape-based style. If you want to do a tape-stop effect then this will pull that off perfectly. And it’s also great for other effects and doing interestingly wobbly things with audio.

Mixer View Although the console looks and acts the same, there’s one new feature that’s potentially revolutionary: Mixer Scenes. In a nutshell it’s the ability to save snapshots of the mixer with all the plug-ins, sends, panning, levels and routing as a Scene. Perhaps in the past you would have put together a basic mix and then saved a new version of your project so that you had a safe place to go back to if it all went

w w w . s o u n d o n s o u n d . c o m / September 2020

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ON TEST PRESONUS STUDIO ONE 5

You can make all sorts of adjustments with Clip Gain Envelopes.

wrong. Now you can simply save it as a Mixer Scene and come back to it with a single click. You can use it to compare mixes or perhaps try three different plug-in chains on the vocals. It’s great for storing points of progression in a mix or trying out new ideas without getting lost in lengthy Undo histories. It doesn’t have to be everything all at once. You can have it recall just the visibility of tracks so you can hide a group or two to focus on a specific collection of channels. You could create scenes based on Mutes for instantly just picking up the vocal tracks, or the drums, the guitars, etc. It’s useful for diving into some ideas and then being able to come back to those same tracks later on as if nothing happened. It’s so ridiculously useful that you’ll wonder why it hasn’t always been there. I’ve already allocated it a keyboard shortcut and it’s dropped invisibly into my workflow.

Listen Bus Under the Spanner icon on the mixer you’ll find a new feature called the Listen

Bus. This is primarily aimed at users running a studio situation where you don’t want the musicians monitoring on headphones to be bothered by your messing about with soloing tracks in the control room. With the Listen Bus enabled and routed to your main speakers the soloing of a track will only happen on the Listen Bus and not on the Main output that’s feeding the vocal booth, headphones and live-room monitors. This is one of those catch-up features that studio users have been after for some time now and it’s great to see it implemented. Same goes for MTC/MMC synchronisation, which is something you probably assumed was already present like it is in every other DAW, but we’re there now, so happy days.

Aux Channels An option has opened up in the mixer to create Aux Channels. These are channels in the mixer that are designed for the play‑through and monitoring of external audio signals without allocating them a track for recording. So, perhaps you have an external synthesizer and want to be able to hear its output through Studio One. Previously you would create a MIDI track to run the synth and an audio track to hear it. This works fine but can rather clog up your Arrange View with empty audio tracks. The Aux Channel bypasses the Arrange View and gives you a fader in the mixer set to whatever input the synth is connected to. You can make it even more elegant by getting You can save various attributes in a Mixer Scene.

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into External Instruments, which is making something useful out of the New Instrument setup in External Devices. You can create an External Instrument in the Browser for each of your external synths and sound modules and then drag them into your project like any virtual instrument. This automatically creates an Instrument Track routed to your synth and an Aux Channel in the mixer ready to receive the play-through. This greatly improves the ease of connection between Studio One and your external gear.

Score For composers and arrangers the arrival of a Score View and editor into Studio One could be what tempts them over from established DAWs like Cubase and Logic. The Score View exists in the same editor window as the piano roll and drum editor. It’s just another view of your MIDI data available on a click and it is beautifully done. Each track is available as a single or Grand Staff for two-handed pieces and you can toggle each track on or off so you can see it as individual instruments or as a conductor’s score. There’s also a rather lovely option to flip the look to white notes on a dark background. Studio One will have a decent stab at interpreting your playing into some kind of coherent score, but inevitably some editing will be required. Nearly all of the editing tools and features available in the piano roll editor are absent from the Score View. None of the Musical Functions you’d find on the right-click or from the Action menu exist. Your toolset is reduced to an arrow and a pen, which may be all you need and certainly keeps the Score view uncluttered and focused on the task of notation. The arrow tool lets you change the pitch of individual notes and chords but not the duration or placement. You can, however, copy notes and paste

Plug-ins

them about the place by moving the song marker. You can also change pitch with the arrow keys and move forwards and backwards from note-to-note like you can in the MIDI editor, except in the Score editor it selects whole chords together and lets you transpose them. There doesn’t seem to be a shortcut to deleting notes; no modifier that changes the mouse pointer to an eraser and so you have to select the note and press Delete. To change the duration of a note you use the Pen tool and select the type of note

All the included plug-ins have had the makeover they deserve, as many had a ‘version 3’ feel. The Scope plug-in has finally been dragged out of its adorable Windows 98-style interface. The Analog Delay and Chorus have been reskinned into vintage-style effects reminiscent of Roland Space Delay and Chorus Echo. The Rotor rotary speaker gets a wooden cabinet and animated speaker cone visible through a grill. The Tricomp also gets a cool paint job. These are engaging changes that makes them very noticeable in the thumbnailed browser list. The delay gets an exponential volume curve and dedicated Width and Ping Pong controls and all dynamics plug-ins now have side-chain inputs, but otherwise the changes are largely cosmetic except for the noticeable spread of PreSonus’ State Space amp modelling.

from the toolbar, or press 3 to 9 on the keyboard, and click on top of whatever note you wish to change. The score will adapt to accommodate it. If you put more notes into a bar than the time-signature allows they will appear as red and will be ignored on playback.

Wherever there is ‘Drive’ you’ll find the State Space logo declaring that their hi-tech virtual modelling has been used to replicate an authentic drive circuit response. This was first found in the Console Shaper and Fat Channel XT and more recently made it into the reworking of Ampire last year, but now it’s in the Delay, the Rotor and the Tricomp, bringing some extra character. The Pro EQ has been updated to a new version and now includes a phase linear low-cut EQ with fixed values and a variable slope. The spectrum display now has a 12th-octave mode and it has a dedicated input meter with adjustable range and peak hold. I should mention that you now get Melodyne Essentials version 5 included.

You can start with a fresh score and pick up the pen tool to start writing. Choose your note and click it into the empty bars. You can add rests and articulations like ties, trills and glissandos and they all express themselves over MIDI on playback. One neat feature is the

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The new Score view is very handsome and can be seen alongside the MIDI editor and clip.

single command under the Action menu which is ‘Fill with rests’ and will fill in any unused space in the score. The mouse isn’t the only way to add notes. You can enable step-mode and then enter notes of the selected duration with a MIDI keyboard or with your QWERTY keyboard which can be a very fast method of note entry once you get the hang of it. The Score View is a good addition to the editing and composing toolbox and

will be welcomed by the many people who enjoy working with music in this way. It is basic when compared to more comprehensive score-editing software but it’s easy to use and does the job. You can also detach the editor window, pin it and open the MIDI editor alongside to use them together. You might find it quicker, for instance, to do some quantising in the piano roll rather than tidy up notes by hand in the Score view.

Key Switches One small but useful enhancement to the MIDI Editor is the introduction of a Key Switch lane. Key Switching allows a sampled instrument, like those in the Presence XT, to switch between alternative articulations of the same instrument by hitting certain low-octave MIDI notes. You could be playing a legato stringed instrument and then tap C0 on your keyboard controller to switch to staccato sounds. These switch commands are present in the MIDI Editor as MIDI notes but they are often hidden from view because the actual note action is happening in higher octaves. The Key Switch lane makes them visible the whole time in a similar style to the Chord or Arrange Tracks but beneath the MIDI Editor.

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The Key Switch options are available and named automatically when using Presence XT and selectable from a drop-down list. For other instruments like Kontakt you have to create a little Key Switch preset to denote the keys and naming. Studio One also takes the trouble to exclude the Key Switch notes from MIDI processing, so when you apply functions in the MIDI editor or when using MIDI effects such as the Arpeggiator or the Chord Track they don’t get in the way. The Key Switch lane gives you a more elegant way of experimenting with and applying articulations that makes this sort of orchestral workflow so much simpler.

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I’m untroubled by the lack of editing features and the note processing you get in the MIDI editor, but there are a couple of things that are noticeable by their absence. You can’t add chord symbols or text that you think could be easily transferred from the Chord and Arrange track. You can’t view it as a sheet of paper or export it as anything that would be helpful outside of Studio One. This leads you to the realisation that you can’t print it. I guess for that functionality PreSonus are hoping you’ll upgrade to Notion. So, the Score view is really only for those people who would prefer it over the MIDI editor, and that narrows the field rather. However, if you own Notion scoring software or the PreSonus Sphere subscription that includes it (see box) then you can export to it directly from Studio One for professional score printing.

Note Controller Hidden away in the automation lanes and next to the Key Switch editor (see box) is the Note Controller, which is Studio One’s foray into MIDI Polyphonic Expression or MPE. It lets you edit the alarming amount of data produced by an MPE-compatible controller for individual notes.

Select a note and be as expressive as you like with MPE‑compatible instruments.

It takes a little bit of setting up and had me going round in circles for a while. First of all your controller and instrument needs to be MPE compatible and then you have to enable MPE in the setup for the controller and for the External Instrument or Virtual Instrument that you’re hoping to express yourself with. None of the PreSonus instruments appear to be MPE enabled, so I installed the ROLI Studio Player, which is full of MPE sounds, but sadly Studio One doesn’t show you the ‘Enable MPE’ option in the place where you are supposed to enable it. After putting the issue to PreSonus they said that currently only VST2 instruments are supported for MPE whereas I was using VST3. A future update will correct this. I found that Kontakt does give you the option to enable MPE, but I’m not aware of having any compatible content for it. So stumped for virtual instruments I dug out the little Modal Skulpt synthesizer which has recently become MPE compatible and, once enabled, in the ModalApp it worked perfectly. Within the Note Controller lane you can edit Pitch, Pressure and Timbre for each individual note. After a typical squeezing of the ROLI Seaboard the editor is covered with multiple graphs of overlapping expression and PreSonus haven’t done anything particularly innovative here to make it any less daunting. You can choose an individual pitch from a drop-down menu and then draw in the three controllers and they will have an effect any time that note sounds. Or you can select the note itself in the MIDI editor and make your adjustments that way. One annoying workflow quirk was that when you have the paint tool selected to edit the controller data and move the mouse up to select the next note it auto-switches to the eraser tool and deletes it. On the other hand, one great workflow attribute is that you can use all of the useful automation lane paint tools, so you can draw sawtooth modulation on individual filter expressions or sine waves for polyphonic vibrato and transform the data all over the place. It works well enough if you’re prepared to put in the time and it’s certainly very welcome with the potential of MIDI 2.0 looming.

Show Page The Show Page is a lot like Mainstage, Cantabile or GigPerformer. It offers a way to set up a live performance rig so that everything is managed and connected through Studio One in terms of virtual instruments,

live instruments, effects and backing tracks. You could already do this by creating virtual instrument channels and audio channels in regular Studio One and then add effects in the console and save it as a project. But that’s not very efficient and it takes time to load different projects for different setups for different songs. Show Page streamlines all of that into a focused, gig-ready, song-management live performance environment.

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ON TEST PRESONUS STUDIO ONE 5

The Show Page has your gig mapped out and managed.

You’ve got to think of it in terms of a gig. So first you add Players, of which there are three types: Backing Tracks, which are essentially audio tracks; Real Instruments, which are set up as an AUX track for your guitars, vocals and synths, etc; and Virtual Instruments, which are for PreSonus instruments and VSTis. Next you add a setlist of the songs you’re going to be performing and these then appear in the timeline as big blocks of stuff. You can set various attributes like the length, tempo, key-signature and what happens when you get to the end of that song. You might want to plough straight on with the next song or you might want to pause for applause before continuing. You can put in a set pause or only advance when you’re ready. You can also loop songs back to the start, which opens up the possibility of breaking them up into verses and chorus and deciding on the number of repeats on the night. Once your setlist timings are mapped out you can then start setting up the sounds. Each track has a ‘patch’ system which saves the inputs/outputs, loaded instrument preset (if it’s a virtual instrument track) and any effects and mixer settings. The Studio One console is available within the Show Page and the patch system incorporates a similar idea to Mixer Scenes. You can have a different patch for each song in the setlist and

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as the timeline travels from one song to another it automatically calls up the presets, effects and mixer settings. It’s quick, smooth and completely brilliant — your whole gig mapped out and there’s none of that frantically trying to find the right preset in the dark business.

Perform At this point you’re thinking that the Show Page has nailed it, but it’s not done yet. There’s a provocative button called Perform that when clicked throws up a minimalist black screen with a simple mixer controller, master level meters and the setlist: it’s initially a bit mystifying. The idea is to provide you with a distraction-free environment for when you are actually performing, with only the information and

settings you need to pull it off on the night. The mixer controller can be configured with 16 knobs, faders or pads or eight of each. You can then map those to any parameter in the Show Page, whether that’s mixer elements, effects or virtual instrument parameters. And these are Macro controls so you can layer up a whole load of parameters to a single knob or fader. And if you have a touchscreen-enabled laptop then the whole thing is multitouchable, or you can map the controls to an external MIDI controller. You’ve then got your setlist, your patches, timeline, macro controllers and level metering all on one simple screen that stands out a mile in a darkened room. And you can control the whole thing from the faders on your MIDI

The Perform view gives you an overview and macro controls over the entire show.

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controller, or by touch, or by the Studio One Remote app that’s been updated for the Show Page. That’s phenomenal. What’s missing from the Show Page is any way to capture your performance. Being able to record your gig, even as just a mixed stereo file, feels like a no-brainer and multitrack recording can’t be hard for what is ostensibly a DAW. I asked PreSonus about it and the official line is that their solution for live recording is the Capture software, but unfortunately that only works with their Live desks. Their assumption is that not everything will necessarily be running through the Show Page which is true enough, although I’d argue that they are underestimating its appeal. There’s another aspect that occurs to me. If I could record everything going through the Show Page then it could evolve from being just about live performance to being about capturing live jams as well. You could have your modular synthesizer rig wired through it and take advantage of software effects and mixing and record a mix of what you’re doing while remaining essentially DAW-less. You could store MIDI patch changes on your synthesizers for different songs you’re working on without dipping into the internal

sequencer. It’s the sort of feature request that might get a lot of traction.

Conclusion PreSonus have paid attention to their diverse user base and brought in a superb range of new features. I think the effect of the Mixer Scenes on your workflow will be huge, while the plug-in tweaks are nice and the Aux channels are useful. The MPE editing is a timely and important development, although some included MPE-compatible instruments would be good. While the Score view is beautiful and competent I feel it lacks the one thing that would make it appeal to the broadest

range of users. The Show Page is excellent. It’s well thought out, sophisticated and could spell the end for other gigging software, but it could be more ambitious in capturing those moments that only a live situation can produce. The majority of my criticism is about things I feel PreSonus could have done, whereas if I focus on what they have achieved then there’s a huge amount to be pleased about. It’s a tremendous update and the other things may come along in time. ££ £344 including VAT. WW www.presonus.com

Z E N T OU R SYNERGY CORE

PreSonus Sphere We all enjoy a new way of paying for our software and PreSonus does this with their ‘everything bundle’ they are calling PreSonus Sphere. You get all their software, all the premium plug-ins and all the extra content. You also get access to their new online collaboration tools and 30GB of storage for shared projects. Studio One lets you export stems and mixes directly to ‘Workspaces’ you create in your Sphere account which can then be shared with other users. They throw in some exclusive content, member exchanges, videos and training and even give you a direct chat line to the experts for support. How much? It’s £12.14 per month or £133.92 a year. Compare that to Pro Tools at £25 a month or even something like Adobe Creative, which for around the same monthly fee only gets you Photoshop and Lightroom. The pricing is impressive, achievable, realistic and it keeps you in the upgrade loop. The full version of just Studio One Professional to buy and keep is £344.40 and you’d have to pay for any versions upgrades that come along.

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ON TEST

Plugin Boutique Scaler 2 Scale Detection & Generation Plug-in

Scaler 2 is just the thing to get you out of a songwriting rut. JOHN WALDEN

A

little music theory can be really useful in the compositional stage of a project, and those who have not passed Music Theory 101 might find a bit of software assistance to be invaluable! I reviewed the original version of Scaler in SOS December 2018 — www.soundonsound.com/reviews/ plugin-boutique-scaler — and, as the core features and much of the UI remain intact in v2, my main focus here will

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be on the significant new features and refinements. However, a brief overview might be helpful. Scaler is built around an extensive knowledge base of chords, scales and harmony. The UI provides access to this via three broad zones: the upper ‘detection’ zone shows chords that are identified in incoming MIDI; the middle zone shows either the scale/chord combinations that best match those detected above, or a scale/chord combination from one of Scaler’s many presets; and the lower zone lets you experiment with chord sequences. For auditioning, you can trigger even complex chords with a single MIDI key, monitoring via a very usable selection of internal sounds or any virtual instrument in your DAW. Some MIDI keys are

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Scaler 2 will look familiar to existing users, but there are lots of new features to be found in the GUI.

Plugin Boutique Scaler 2 £44 pros • A music-theory assistant in a plug-in. • Excellent new performance and key modulation options. • New audio-detection facility. • Cracking value for money.

cons • None.

summary This highly impressive update to Scaler boasts some excellent performance options and expanded music-theory support for composers. It’s also an absolute bargain.

Pad mode allows you to keyswitch between multiple Patterns (chord sequences) to experiment with your overall song structure.

mapped to specific chords, but the rest of the note range can be used for melody creation alongside your triggered chords, and the notes can be constrained to fit the selected key/scale (ensuring no duff notes). Whether just as simple block chords, or as a ‘performance’ you create from the mapped chords and scale-corrected melodic key range, the

MIDI can be copied to a DAW track for any further manipulation and playback via a virtual instrument of your choice.

Chord Explorer So, what new features does v2 bring to bear on your exploration of which chords might or might not work in your composition? Scaler v1 already installed

multiple plug-in versions to cater for the routing options of different DAWs and plug-in formats, but an impressive new version caters for a new audio detection feature. Yes, Scaler can now attempt to detect chords/scales from audio, not just MIDI. On the whole, this works pretty well when used with a single instrument (keys or guitar, for example). It’s perhaps not as foolproof as the MIDI detection process, which is no surprise, but it’s a great addition nonetheless. There are some extensive additions to the Songs and Artists chord-set presets, too. These were always a good source of inspiration for new musical ideas, and the

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ON TEST PLUGIN BOUTIQUE SCALER 2

extra options have plenty to offer for both songwriting and media music applications. Also useful are the improvements in the lower zone, where it’s now much easier to create multiple chord Patterns (sequences) in a single Scaler instance. This allows you, for example, to develop full intro, verse, bridge, chorus and outro sections. This section of the UI can now be expanded to a ‘pad’ view, showing your multiple Patterns in a single screen. When in this mode, you can switch between Patterns via keyswitches, making it easy to experiment with different song/composition structures without leaving Scaler.

That’s The Way When it comes to ‘performing’ with your chosen chords/scale combination, Scaler 2 is much improved in a number of ways. For example, there are now additional chord voicing and chord variation options in the central section of the interface, providing you with a range of different flavours and/or chord-substitution suggestions. The highlight, though, is the new Perform panel. There are lots of useful features, including a substantial collection of performances, phrases and rhythms. In essence, when you trigger a chord via a MIDI key, the performance presets provide unique playback patterns — for example, arpeggios or rhythmic triggering of the full chord. The panel includes humanise options for velocity and timing, and a voice-grouping feature, which essentially provides various chord-voicing options. For example, the ‘dynamic’ option provides you with a bass note and automatically identifies inversions of each chord, to keep their fingering close to the scale tonic. Other options provide ‘grouping’ to force chord playback into particular octave/note ranges, which is useful when composing for instruments with a limited note range (some orchestral instruments, for example). Wonderfully, if you engage the Edit mode in the lower ‘Pattern’ section, the performance options can be specified on a per-chord basis. You have control over the octave, timing and performance details that will be applied when you trigger that chord. When you then hit the MIDI Capture button, all these details are included in MIDI data that can be dragged to a DAW track. Finally, Scaler’s Key-Lock options, which allow you to play harmonically

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correct melodies in the upper regions of the MIDI note range, has two new choices: Chord Notes and Chord Extensions. These map just the chord tones, or the chord tones and some additional notes, to the upper section of the keyboard and provide further, ‘duff-note-free’ melody possibilities. Version 2 improves the options for saving Scaler presets to, making it easier to copy a complete configuration to other instances of the plug-in, and to other projects. Even better are the new ‘sync’ options, that allow selected settings to be synchronised between different instances. For instance, you might have three instances of Scaler controlling piano, strings and brass instruments, all configured with different performance settings for the same underlying chord sequence. However, if you edit the chord sequence for a specific pattern (ie. song section) in one instance, rather than have to make the same edits in all instances, you can ‘sync’ the changes between the various plug-in instances; it makes for a much slicker workflow.

Mods & Modes Another highlight is the modulation system, which I’ve found very impressive. Accessed via the Modulation button in the central section, this allows you to explore all sorts of possibilities for modulating between keys, and offers advice on which chords might form a good pathway to do that. Modulation can be achieved in a number of different modes. For example, starting from your current key in the default Progression mode, you can select a destination key via the Circle Of Fifths graphic. Scaler will show your current chord progression transposed to the new key, and provide a series of chords (sometimes with multiple options) that provide a pathway from one key to the other. Progression mode is great for simple key changes when you’re songwriting, but the other modulation modes are well worth exploring in other situations. For example, the Mediants option could be very cool for media/film composers. It encourages you to explore creative routes between chords that lie outside the currently selected key and can

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The Perform panel brings a whole new dimension to triggering your selected chords.

suggest some really interesting chord changes/sequences that those without a music theory PhD might not find by semi-random experimentation alone. It’s very clever stuff.

Scaling Up? Scaler was already an extremely useful utility plug-in, but Scaler 2 takes the experience to another level. The audio detection is certainly a useful addition but, for me, the absolute highlights are the new performance options and the impressive modulation system. The first of these gives Scaler a touch of the ‘performer’ element found in Toontrack’s EZkeys. Not only do you get help with your chord selections and melody writing, but you now also get some interesting and expressive performances of them. The modulation options might be something that really only interest the more musically adventurous songwriter or composer, but they can also encourage those with a somewhat sketchy grasp of music theory to explore what’s possible. Scaler 2 is a bit of a triumph, and at this price, whether you’re buying new or upgrading, it’s an absolute steal for those wanting to expand their compositional toolkit. ££ Scaler 2 £43.73; upgrade from Scaler 1 £16.96. Prices include VAT.

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ON TEST

Rupert Neve Designs RMP-D8 Eight-channel Dante Mic Preamp This remote-controllable preamp boasts some unusual features aimed largely at the live-sound market. HUGH ROBJOHNS

A

udio-over-IP (AoIP) continues to gain popularity, particularly in live-sound and broadcast applications, and Rupert Neve Designs have recently joined the network-audio crowd, with the release of the RMP-D8. This eight-channel, remote-controllable microphone preamp incorporates the almost ubiquitous Dante AoIP interface, and although aimed primarily at live-sound applications, it offers exemplary audio quality and simplicity of use for any Dante-equipped installation, so should also find welcoming homes in the

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broadcast, education, studio-complex and location-recording sectors.

Controls & Construction The RMP-D8 looks nothing like any other RND product I’ve seen. With its sturdy 2U rackmounting steel chassis and control panel, both finished in plain matt black, it’s far more industrial-looking than the Portico or Shelford ranges. Ten vertical slots milled into the substantial front panel reveal eight digital bar-graph meters and two LED arrays of status indicators. User controls comprise 15 illuminated buttons for channel access and configuration, a big red indented

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rotary encoder, a black power button, and a small colour OLED display. All the usual preamp facilities are provided for each channel, with individually switchable 48V phantom power, polarity inversion, a 10dB pad, an 80Hz (12dB/octave) high-pass filter, and gain that’s adjustable between 0 and +60 dB in 1dB increments. If the line input is selected, the gain range is restricted to a maximum of +30dB, phantom power is automatically disabled, and the pad is engaged. A Reset button restores all channels to their default settings. When one of the channel buttons under the meters is pressed, that channel’s settings are shown on the OLED screen. The preamp’s gain is displayed in large numbers, along with the status of the phantom power, high-pass filter, pad and polarity options. Also shown is whether gain compensation (GC) is active (more on that

later), and whether the front-panel controls have been locked. Additional menu screens show system information like the current firmware version, PSU voltages and temperature, the Dante connection IP/Mac addresses, Dante software and ID details, and the device ID for use with remote control from Yamaha CL, QL or PM consoles. The preamp functionality is obvious and efficient, making routine setups and operations straightforward, but the 20-page manual is necessary reading if you’re to understand the more involved Dante configuration functionality. The only thing I felt was missing was an instant overview of all channel settings; the assignable nature of the controls means that the unit’s front panel can’t provide that overview, so I’d very much like to see an overview display page added to the OLED screen in a firmware update. The RMP-D8 features true mains power redundancy: there are two separate universal voltage (100-240 V AC) power modules, each fed from separate mains supplies, with their outputs switched automatically if one supply (or PSU) fails. The two provided mains cords are fitted with locking IEC plugs, too, so there’s zero risk of the cables falling out after being bounced around in a wheeled

rack or an OB truck. These provisions are important for applications where the equipment serves in ‘mission critical’ or arduous roles. There’s a cooling fan, but it’s temperature-controlled and quiet: on the occasions it decided to run I could barely hear it, even when right beside me on the desk. The eight mic/line inputs are connected via a single row of combi XLR/jack sockets on the back panel, and phantom power is never present on the TRS connections. I was surprised at the omission of a multichannel input connector; there’s plenty of space for a couple of AES59 D-sub connectors for separate mic and line inputs, for example. Another surprise was that there are no analogue outputs: no line-level preamp outputs, and no headphone socket for local monitoring, which would have been useful for checking connections and settings during rigging, or when fault-finding. There are four male XLRs carrying AES3 digital outputs from all eight channels, although I’m struggling to imagine what these would typically be used for. Also on the rear panel is a USB-A socket for firmware updates, and a pair of RJ45 sockets providing primary and secondary Dante network connections.

Signal Path As you’d expect of a Rupert Neve Designs product, the analogue circuitry is described as Class-A and each channel features a custom-designed transformer. These are not the mic input transformers you might have expected, though; the front end is actually electronically balanced with an ‘instrumentation amplifier’ circuit topology. Instead, the transformers reside in the channel output stages feeding the A-D converters, and are credited for delivering the ‘larger than life’ tone so often associated with Rupert Neve’s preamps. Apparently a lot of attention has been paid to the RMP-D8’s digital clocking and jitter-reduction arrangements, and its 24-bit converters support all the standard sample rates between 44.1 and 192 kHz. Unusually, the four common pull-up/pull-down sample-rate modifiers required for some film/TV shoots are also provided (±0.1, -4 and +4.1667 percent). As a pure Dante interface, the desired sample frequency can only be adjusted via the Dante Controller application; there’s no provision to adjust the sample rate locally.

Dante & GC Outputs Although an eight-channel preamp, the RMP-D8 actually presents 16 outputs to

the Dante network. The main outputs are labelled channels 1 to 8, as you’d expect, while channels 9-16 can either be straight duplicates (mirrors) of the first eight channels, or ‘gain compensated’ (GC) versions, selectable on individual channels. This ‘GC’ refers to a feature whereby the channel gain can be adjusted for the output without affecting the previously set gain structure. For example, if Channel 1’s gain is turned up, the GC output has its gain turned down automatically to compensate, thus maintaining a consistent level at that output. The GC outputs always start with 6dB less gain than the main outputs, to build in some working headroom, and the automatic gain compensation only works over a ±12dB range, and within the limits of the preamp’s 0-60 dB overall gain range. Why is this facility provided? Well, in a typical live-sound setting the main (1-8) outputs would typically be routed to the FOH console and the GC (9-16) outputs to a stage monitor console. During the soundcheck, let’s say channel 1’s gain of +36dB is stored as the initial setting, and the gain to the GC output will therefore be +30dB. During the show, the FOH engineer wants a bit more level and tweaks the gain up to +40dB, the GC output to the monitor desk will still have 30dB of overall gain, so the artists’ monitoring feeds don’t change either. While the FOH engineer could use the front-panel encoder to change channel gains, the RMP-D8 can also be controlled

Rupert Neve Designs RMP-D8 £5499 pros • Dual redundant mains supplies. • Classic larger-than-life sound character. • Very solid technical performance with generous headroom. • Remote-controllable and compatible with Yamaha’s RIO system. • Provided with locking IEC cables.

cons • No display overview of all channel settings. • Fixed-speed encoder makes setting high gains tedious from the front panel. • No analogue outputs or local headphone monitoring.

summary Optimised for professional live-sound AoIP installations, the RMP-D8 provides a high-quality eight-channel, remote-controllable preamp with full Dante connectivity.

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ON TEST RUPERT NEVE DESIGNS RMP-D8

Two separate power supplies, each with its own IEC inlet, provide true mains power redundancy.

remotely over the Dante network; RND provide free Mac/Windows remote-control applications that run alongside the Dante Controller and it’s possible to control the preamp channels directly from any Yamaha CL, QL or PM digital console (the RMP-D8 appears as a native Yamaha RIO preamp, allowing control of gain, phantom power and high-pass filter settings). RND’s remote control app provides a graphical representation of the eight channels, with gains being adjusted either via on-screen knobs or by typing in numerical values, and switch functions are toggled by clicking the on-screen buttons. Setting up this interface is a bit ‘geeky’, but it works well. Although the app version I used during the review could only control one physical RMP-D8, RND were already beta-testing an updated version which can address eight units simultaneously, and that should be available by the time you read this.

Performance As you might expect, the RMP-D8 boasts some very solid technical specifications, all of which I confirmed with an Audio Precision test system. The mic preamp EIN figure (150Ω source, 60dB gain, 20Hz-22kHz bandwidth) measured the equivalent of -127dBu (-92.5dBFS at the AES3 output), and the system noise floor at unity gain was -100dBFS. The AES17

dynamic range figure measured a tad over 115dB (A-wtd), putting it on a par with the Audient ASP880 and Antelope Orion. The THD figure was 0.007 percent with an output level of -0.5dBFS, showing a slight emphasis of odd harmonics over the even ones, thanks presumably to the output transformer. The frequency response is easily within ±0.25dB from about 30Hz up to just below the Nyquist frequency, and I measured the low-frequency -3dB turnover point at 10Hz, rising to 80Hz with the second-order high-pass filter engaged. The input impedance is a Neve-classic 5.3kΩ for both mic and line modes and, unusually, it doesn’t change if the input pad is activated. The unloaded phantom power voltage was fractionally low, but comfortably within spec at 46.9V, and it delivered 10.8V at the microphone when providing the maximum 10mA current, which is also safely within spec. The maximum input level corresponding to 0dBFS (digital clipping) is +25.5dBu, while the SMPTE specification calls for +24dBu, so RND have effectively built 1.5dB of extra headroom into the conversion. (While this is potentially useful in a live-sound situation, it means a standard +4dBu analogue reference doesn’t generate the expected -20dBFS digital output. Thankfully, only engineering geeks like me would fret about such trivial anomalies!)

Alternatives There are now several octal Dante mic preamps, and some of them share a Rupert Neve heritage. These include Neve’s new 1073OPX and Focusrite’s older ISA828 MkII (which can be equipped with a Dante interface, but it isn’t remote-controllable). Focusrite’s Rednet MPR8 is remote-controllable, though, and also has dual-redundant power supplies. Other remote-controllable eight-channel Dante preamps include Glensound’s DARK8MAI, Grace Design’s m802 and Millennia’s HV-3R.

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In Use Once the RMP-D8 is integrated into a Dante network, it’s very easy to use both from the front panel and remotely. The bar-graph meters show -60 to -9 dBFS across five green LEDs, with two oranges for -6dBFS and -3dBFS followed by a (helpfully pessimistic) red overload LED. (I like a little more warning of high peaks, and would have preferred the -9dBFS LED to be orange.) Unfortunately, setting

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high gains from the front panel can be rather tedious, because the encoder isn’t speed-sensitive — a lot of revolutions are needed to reach +60dB. The controller app is a lot faster in that respect. I also couldn’t find a way to pair channels for convenient stereo operation. Although the RMP-D8 is intended to be used with a Dante network connection, it can be used as a standalone preamp/ converter under certain circumstances. The unit retains in memory its Dante-programmed clocking configuration and restores that setting when the power is cycled. So if the unit is connected to a computer running the Dante Controller and programmed to use it’s internal clock, it retains that capability afterwards, allowing use as a free-standing preamp with AES3 outputs. It’s not particularly convenient, but it works! Sound-wise the RMP-D8 is great, being quiet and generally clean, with more than enough gain for most live-sound applications. The standard facilities are effective and working with normal headroom margins delivers a sound which is full-bodied and does have a slightly larger-than-life character associated with well-engineered mic preamps. It’s not obviously coloured, but intentionally driving signals into the red results (initially) in a progressively saturated distortion rather than the more typical hard aliased clipping. Although I feel there are a few areas where some minor firmware updates would be beneficial, overall this is a good Dante preamp and should do well in the AoIP live-sound market for which its unusual feature set has clearly been focused. ££ £5499 including VAT. TT Rupert Neve Designs UK & Europe +44 (0)208 191 0058

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TECHNIQUE

Tips For Keeping Musicians ‘In The Zone’

NEIL ROGERS

A

t some indefinable point in the past, I completed my transition from drummer to engineer, and while I still lay down parts when clients need them, I stopped thinking of myself as ‘a musician’ long ago. But last year I was invited by a client (who I’d played drums for on one track) to drum on their latest album. It was a useful experience: being in a different studio (StudiOwz in Pembrokeshire), with someone else engineering, reminded me what it is about recording sessions that helps or hinders performers. With that experience in mind, here are 10 ways I think engineers and studio owners can ensure the studio works as well for the musicians as it does for themselves.

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Time & Place Recording sessions can be long. If working with a band you might not need all the artists there simultaneously, and they might or might not want to hang around while you’re recording their bandmates. You can’t plan a session down to the last minute — recording doesn’t work that way — but if you know you’ll be setting up the drum sound all morning and the guitarist and vocalist are free to do what they want until early afternoon, let them know that, and tell them when they need to be back.

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Communication Mics You’ll have a talkback mic in the control room, but how will the artist talk to you? It’s easy enough with a vocalist (use the vocal mic), but you don’t want other performers to have to keep repeating things because their voice is faint, or to shout, or stretch to reach a mic (potentially moving things as they do!). So have a listen mic or two set up in the room which can pick up what the artists are saying. An instrument or room mic might double up for this purpose, but with a dedicated mic you can add a compressor to make it easier to pick up artists from further across the room. Just be sure not to feed this signal accidentally into the cue mix! Record it, and as a bonus you’ll have a record of their reaction after each take.

An Inspiring Cue Mix

Manage The Pace Just because you can do things fast doesn’t mean you should. You certainly don’t want to move so fast all the time that the session feels hurried. A good engineer should know when to suggest a little break, how much time to spend indulging an artist’s new idea, and when to slow or quicken the pace more generally. It’s about finding the rhythm for a particular artist or session that helps get the performance in the bag. So learn your tools and prepare in advance: that way, you can choose when to work quickly and when to back off the pace. You’ll be in control, and you’ll also be seen to be in control, which will boost everyone’s confidence.

What an artist hears in their headphones has an impact, and engineers should strive to make what the performer hears inspire them to deliver a great performance. Modern tools make it easy to set up decent monitor mixes, so take the time to learn that side of things like the back of your hand. But also try to understand what sounds and effects in the headphones will really help. For the StudiOwz drum session, I was playing a great-sounding kit (so great I bought it!) in a great-sounding room. Though you can hear drums while playing, they always sound different at the playing position to out in the room. The engineer made a point of feeding some of the room mics into my cans. Knowing that my performance sounded good in the room gave me a real boost, but these mics also gave me a better sense of the internal balance and the depth and power of the kit, which helped me to avoid overplaying. This advice isn’t unique to drums. A lot of guitarists seem more comfortable recording in the live room with their amp and effects than playing to speakers in the control room. Committing to sounds created with pedals or outboard equipment can terrify new engineers, I know, but recording isn’t just about capturing neat and tidy files that you can create something with at the mix stage, so it can be worth taking the ‘risk’, and you can always capture a dry insurance signal. When it comes to vocals, having a few obvious effect options to hand for the monitor mix can be useful. A slap delay and a short reverb can often help a singer get into things.

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TECHNIQUE T H E A R T I S T- F R I E N D LY S T U D I O

Artist-operated Cue Mix

Photo: Andrew Stawarz

Some musicians like to have control over their monitor mix. I certainly do. The most basic requirement is to be able to set the volume in my headphones, but ideally I’d have a simple, clearly labelled headphone mixer that allows me to balance just three signals: my own performance, all the music and the click track. That way, I can set a comfortable balance without having to discuss every detail with the engineer. With some musicians, there’s only so much back and forth you can do before they start to feel uncomfortable asking for yet another small change. Sadly, headphone amplification is one of the corners most often cut when setting up a studio. There are a few ways to set up multichannel monitoring systems, ranging from bodge-jobs with small mixers, to smartphone-controlled systems and dedicated multichannel monitoring systems, such as those by Hear Technologies and Aviom. Whatever you can afford, make an effort to ensure your headphone monitoring setup is as good as possible.

The Environment Anything that contributes to a relaxed atmosphere and a good ‘vibe’ will have a much bigger impact on most clients than your collection of mic preamps. Some studios will benefit from being in a stunning location or amazing building, but I reckon any studio can be made inspiring and welcoming in its own way. My studio is in a warehouse on an industrial estate, so the approach will never be awe-inspiring, but I’ve put time and thought into making it feel nice, comfortable and welcoming inside. Lighting, artwork, seating, refreshments, somewhere to hang your coat... This stuff is really important!

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TECHNIQUE T H E A R T I S T- F R I E N D LY S T U D I O

The Need For Speed

Timely Feedback

Modern DAW software leaves us with no excuse for being unable to hit record when needed, and as a performer, it’s really frustrating having to wait for an engineer to cue up a section for a drop-in, or while they create a new playlist. So prep your DAW sessions. Use templates. Learn the shortcuts for creating playlists. Make notes. Insert markers in your arrangement window and assign them shortcuts. Also, make sure you have a lyric sheet handy: a lot of artists like to use lyrics to indicate which part of the song they want to punch in from. (It will also help you spot mistakes!)

What you say to a performer after they’ve finished a take can have a huge effect on their mood. Here’s a great piece of advice I received early on: whatever you have to say, say it as soon as the take has finished, because if a musician has poured their heart into a performance but has any doubts about it, being greeted with silence can be really disheartening. What you say depends so much on your personality and how well you know the artist in question but you don’t have to be George Martin, or say anything deeply profound. As a minimum, make sure the performer has the sense that you’re interested and positive, and understand what they’re going for. I often like to try and distract a performer by asking some questions about themselves; people generally like others taking an interest in them and it can help them relax. It can also help a player mentally ‘reset’ if they’re becoming stuck. Something else I noticed in my drumming session was that I couldn’t be seen from the control room. I quite liked that, as I felt free to do whatever I wanted to get myself ‘feeling’ what I was doing, but this arrangement makes communication over headphones even more important.

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Keep Playing

Keep The Buzz Alive

Increasingly, the type of production projects I work on tend to be less about simply recording a band, and more about helping an individual or a smaller group of people develop and capture their sound. Often, this means getting ideas down quickly. And if you, the engineer, can also jump behind a drum kit, play a bass line, or lay down some keys or backing vocals, that’s really helpful. So when you have a bit of downtime, sharpen up your old playing skills or learn some new ones. It will make you a better engineer, and you’ll end up getting better performances down ‘on tape’.

Making musicians wait while you fiddle can ruin a session. With my engineering hat on, I could happily spend hours fine-tuning the kit and placing mics, but if the drummer has come into the studio feeling pumped about recording their latest track, hours spent hitting individual drums and playing the same loop will really take the edge off. So do as much of the setting up and fiddling as you can before you invite the musicians in to play. For drums, the engineering side can be particularly time-consuming, so if the drummer’s bringing their own kit, have a plan for when they should bring it and set it up. If you have an assistant, you can cut out a decent chunk of the buzz-killing soundcheck by getting them, not the drummer, to hit individual drums. The same applies to other performers: the less they have to help you, the more ready they’ll be to deliver when you hit record. Make sure you have enough mic options to hand for a vocalist. If using multiple mics on a cab, get them in phase and in a great place to start tweaking. Test your cables regularly so you don’t waste time fault-tracing and swapping them out.

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ON TEST

Yamaha CP88 & YC61 Stage Keyboards

Yamaha look to reclaim their stage keyboard crown with two world-class live instruments. ROBIN BIGWOOD

Y

amaha’s stage keyboard heritage goes back more than half a century, and in the 1970s their CP-series electro-acoustic pianos and YC combo organs were amongst the most useful and desirable instruments available to gigging keyboard players. After a long hiatus the CP-series was revived a decade ago, and the CP4 Stage (which I reviewed in the February 2015 issue of SOS) is often cited as an ideal stage piano of the modern era: musical, practical, versatile and musician-friendly. It’s not entirely clear yet whether the CP88 on test here is a complement to the CP4 or a replacement, but either way it’s a new interpretation of the stage keyboard concept for Yamaha.

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And now the YC series makes a storming comeback too. Well, we already had a sniff of it in the form of the little mini-key Reface YC of 2015, but the new YC61 is a different and bigger beast altogether.

Red Or Dead? We’ll dig into the detail of each individual keyboard in a moment, but let’s first consider what unites them. To start with I’ll just come out and say it: these Yamahas are really Nord-like! If you’ve even a passing interest in stage pianos and Hammond emulators you’ll know that the Nord Stage, Piano and Electro are huge players in this field. The Swedish way has always been to make most aspects of sound selection and control knobby, tactile, direct and

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responsive, which is a far cry from some Japanese stage keyboard designs of the last two decades (not least the CP4) which have voluminous hidden feature sets in the depths of menu systems. A Nord-like directness is exactly what we see here. Although both keyboards rely on 128x64 dot LCDs and a menu system for many tasks, all the individual sound-generating sections are equipped with chunky knobs, rocker switches, encoders (with LED surrounds) and two-digit displays. Particularly distinctive are the retro silver metal switches near the keyboard: these toggle a section’s on/off status when you momentarily push them up. The overall design gives the impression of interacting with lots of single-task discrete devices, very much in keeping with the current vogue for all things analogue and/or modular. Another leaf out of the Nord book: these new keyboards are chips off the same block, clearly part of the same wider family. Casework is all metal, high

Yamaha CP88 & YC61 £1729 & £1599 pros • The YC61’s Hammonds are state of the art. • The CP88’s rich piano complement is perfectly matched by a refined and renowned hammer action. • The knobby interfaces invite experimentation. • Great-sounding effects and lots of them. • Sophisticated configuration options and good memory management. • Looks like they’d survive a nuclear war.

cons • The CP88’s non-piano sound provision is sparse. • YC61 transistor organs are far from being faithful recreations. • Panel controls are not always as intuitive or powerful as you’d wish for, and menu-diving is still required for many tasks. • No aftertouch.

summary A pair of pro-level gigging keyboards that represent a significant departure from typical Japanese stage piano design, combining strong piano and tonewheel organ capabilities with an inviting knobby and somewhat retro user interface.

quality, and really confidence-inspiring. The heavy-duty vibe extends out the back too, with sockets nutted firmly to the casework and proper 3-pin IEC mains inlets. The CP88 weighs 18.6kg and the YC61 7.1kg. There are big overlaps in sound-generating technologies and soundsets too. Yamaha say an Advanced Wave Memory AWM2 engine is used for everything but the YC61’s organ sounds, but the pianos have the seamless velocity response behaviour of an acoustic modelling system, and there’s plenty of FM knocking about too. Seamless sound switching is evident, allowing presets to be dialled in without silencing anything already sounding. The implementation looks robust except for an edge case where the YC61 VCM (Virtual Circuit Modeling) organ section changes mode, and that is perhaps why Yamaha don’t trumpet this ability in its marketing blurb. So this is all a bold and interesting design departure for Yamaha. Let’s see how it plays out in practice.

YC61 Although smaller than the CP88, and at first glance simpler, the YC61 is arguably

the more sophisticated of the two models on test here. It’s cheaper than the big hammer-action board, but not by much. Centre-stage goes to a fully featured organ emulator, which is supplemented by a broad base of gigging sounds with simple editing features. The 61 semi-weighted velocity-sensitive waterfall-action keys are fast in action and perfect for slippery organ playing, palm glisses and fall-offs. Yamaha employ its slightly narrower 160mm octave width here, which I personally don’t notice, and should concern only a small subset of players very sensitive to it. There are in fact three separate, independent sound generators on board: the VCM/FM Organ, plus two identically equipped ‘Keys’ sections, A and B. Tied to these are no fewer than nine effects processors of one type or another. The Organ has a dedicated preamp drive, and each of the Keys its own pair of multi-effects in series. An additional multi-effect is on hand to be applied to a single section of your choice and a Speaker/Amp simulator naturally fulfils Leslie speaker duties, but can just as easily dirty up your electric pianos. A simple Reverb is shared by

all three sound generators in a send/ return arrangement, and finally there’s a master-level EQ.

Organic Reach The YC61’s organ section is really versatile. On the Hammond front you have the choice between a ‘standard’ model called H1, a more aggressive midrange-driven and electrically aged H2, or H3 with its very pronounced percussion. Beyond this there are three FM-based organs: a clean sine-wave model and emulations of British and Italian (presumably Vox and Farfisa) transistor designs. The other characterful originals from the Reface YC — the Acetone and a ’70s YC — are sadly missing. All of the organ models will work as two organs in parallel, with separate registrations for a notional Lower and Upper manual that can be played from either side of a keyboard split, or from the YC61’s own keyboard in conjunction with an external keyboard controller feeding the MIDI In socket. The Hammond sounds, allied with the Rotary speaker effect, can be staggeringly good. Other manufacturers also get great results these days of course: the best Nord, Roland and Kurzweil tonewheel

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ON TEST YA M A H A C P 8 8 & Y C 6 1

The YC61’s rear panel is a tick list of everything you want to see on the back of a keyboard. IEC power supply: check. Full-size MIDI ports: check. Full provision of quarter-inch pedal inputs and audio I/O: check.

sounds are also hugely convincing. But the sense of realism and responsiveness here is uncanny, and it really can feel like you’re in the company of a large, loud mechanical presence! At full tilt the organs snarl and growl as does a B3 and a cranked Leslie cabinet. All manner of remarkable side-band hums and hisses spill out, in a realistically unpredictable manner, aided by a menu-adjustable inter-tonewheel leakage level. The YC61 will subtly purr and throb away just as readily though, all the while retaining the sense of presence and weight. Vibrato/ Chorus and Percussion behaviour is sophisticated and convincing: I loved the Vibrato 3 setting particularly, which comes with all sorts of electrical degradation, and there’s a menu option to unlink the 1’ drawbar from its default role in percussion generation and liberate it as an independent footage. There’s also an option for an attached expression pedal to affect only organ volume or Leslie drive as well. The physical drawbars have a good level of clicky resistance over their movement. They also get LED backdrops (of user-selectable colour) that indicate registration when presets are loaded and the physical drawbar positions are (inevitably) wrong. Each has a nifty little transparent section that lets underlying LEDs show through, which is an ideal compromise in marrying physical drawbar feel with preset recall, and arguably an improvement on any methods currently employed by other manufacturers. Matching physical drawbars to saved registrations can be done in a couple of ways familiar from the hardware synth world, with options for an immediate value Jump or a more benign Catch.

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‘Half-moon’ controls for rotor speed and brake are on the left-hand side of the panel next to the drawbars, and by default the sprung left-right pitch stick duplicates the slow-fast control. These are in fact merely triggers for the processing taking place in the Speaker/Amp effect section further to the right. There you’ll also find the alternative ‘Rtr B’ Leslie cabinet which is capable of frankly unsettling amounts of throaty transistor distortion, and a further four non-rotary guitar amps whose overdrive and distortion characteristics can be well suited to the combo organs and electric pianos. A Tone knob dials in a ‘smile’ EQ curve with boosted treble and bass to the right and a drier mid-range balance to the left. Now for the FM organs. They’re useful to have... but rather curious. Least contentious of the three is the FM sine organ. As well as some beguiling clean, glassy textures it soon roughens up into a more generic electric sound with the application of drive and speaker simulation. F2 and F3, notionally the Vox and Farfisa, are quirky. F2 has only one drawbar, the 4’, which generates a distinctive square/pulse wave tone, while all others are still sine/FM-like. F3 has an 8’ transistor tone on the 16’ drawbar, a 16’ tone on the 5+1/3’, and a 4’ on the 1+1/3’. Other drawbars are at similarly unpredictable footages and offer very muted transistor sounds alongside tonewheel-like sines. Both these models abruptly lose their buzzy footages above note B5 and also can’t access the Vibrato/ Chorus and Percussion sections, which is both hugely disappointing and (with the majority of Farfisa Compact models in mind) inaccurate. A generic vibrato can be achieved by pushing up the modulation lever, but it’s a weak affair. Essentially, if you’re after spot-on transistor emulations, these are not they. Sounds and registrations are off, and selecting them by Hammond drawbars

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is all wrong anyway. I didn’t have one on hand to directly compare, but everything I’ve heard of the little Reface YC’s transistor emulations seems to trounce what’s here. However, in practice, and especially when distorted through some of the onboard amp/speaker simulations, the musical effect can still be extremely potent and persuasive, and various Doors/ Who/Floyd timbres can be teased out that’ll fool most people most of the time. So they’re not bad organs: they’re just not particularly authentic.

Key Skills Turning to the YC61’s Keys section, this consists of two identical general‑purpose sound generators which can be played solo or layered and split in any combination with each other, and the organ section. I found the panel layout confusing to begin with. With one set of controls for both it’s not abundantly clear at first that there even are two independent sections, and the black-grey graphics that separate generators and effects are subtle to the point of invisibility. With use and experimentation things make perfect sense. Keys A and B each draw from the same set of 139 sounds, which are selected using a combination of four-way category knob and bi-directional rocker switch. In the section they’re only identified by category and number, but in the LCD you get a full sound name. This is not a prodigious soundset, it must be said, but quality is there when you need it to be. Grand pianos include a fully featured Yamaha CFX, the flagship sound of the CP4, as well as a focused S700 and a rather warm, intimate and likeable C7. There’s also a U1 upright, brimming with character and inharmonicity, two CP80 electroacoustics and some ready-rolled CFX pad layers. In the E.Piano category can be found five Rhodes (or should that be ‘Rhodeses’?)

Exits & Entries

of different vintages, quite varied in character, as well as three variations on a Wurlitzer, two Clavinet tones, a single harpsichord, and seven DX pianos. With all the pianos, the velocity response is beautifully implemented, even if it’s not always easy to control from the YC61’s light waterfall keys. There’s no sympathetic resonance behaviour, but a pedal-triggered damper resonance can be inserted in one of the effects slots; not the most naturalistic implementation I’ve heard, but effective enough, and with the potential to be creatively tweaked to a reverb-like extreme. Acoustic pianos are capable of partial-damping and half-pedalling effects, responding well to continuous-type sustain pedals. In the Synth category you get all kinds of pads, stabs, leads and a few basses and bells. Many attractive fizzy Oberheim and Jupiter textures are here, nicely warm and lively, along with typical D50/M1 layers and juicy synth brasses. Some Moog leads would benefit from more brightness in the upper octaves, and whilst all lead sounds can be made monophonic via a menu option (and indeed are almost always presented this way in the factory Live Sets) some subtle envelope retriggering is audible with legato playing that you don’t get on a real monosynth, even when portamento (another menu option) is employed. As for synth-like parameter editing: there’s some, but it’s very limited. Aside from another smile-EQ Tone control there’s just a single knob, switchable between EG (Envelope Generator) and Filter. In Filter mode it tweaks the cutoff of a low-pass filter, against some predetermined resonance curves (selectable in menus, or using a nifty key shortcut). In EG mode the knob adjusts combinations of Attack, Decay and Release at once (ditto). Many factory Live Sets load up with useful knob responses preconfigured, and a lot can be achieved in a couple of

finger twists. Pad sounds, for example, often have the EG knob simultaneously lengthening both attack and release time, to make them ‘slower’ and more languid. Precise and direct control with this macro-based system is full of inconsistencies, though. Though many sounds do what you expect, the envelope phases of others steadfastly refuse to change very much (or even at all) in any of the 11 EG modes. At this juncture it’s worth comparing Nord’s equivalent envelope system on recent Electros, that uses a two-knob combo to intuitively and reliably serve up almost all useful shapes. But then an Electro has no equivalent of the YC61’s static filter cutoff knob, so you win some and you lose some... Sounds grouped into the ‘Other’ category include section and solo strings, guitars and other plucked instruments, classical and soul/pop brass, a few flutes (including a Mellotron), five useful basses, various mallet instruments, and a couple of squeezeboxes. The quality of nearly all these is very high: they’re clean, energetic and dynamic. Actually none is particularly sophisticated, from a programming perspective, with sample zone transitions clearly audible if you go looking for them and very little velocity switching is employed. There’s certainly no fancy articulation switches or legato transitions. But the fundamental quality is high, and various string, sax sections and even a solo trumpet sound superb, and are very playable.

Effects I mentioned there are two multi-effects processors for each of the keys sections, and the quality of these is in keeping with that of the soundset. What a difference it makes too, to have effect parameters under knob control, even just the two per effect here, rather than buried in menus. Multiple chorus, flanger and phaser algorithms (all great-sounding, and

In keeping with all the under-the-hood niceties, these new Yamahas are well equipped to meet the outside world. Alongside the stereo quarter-inch line out sockets the CP88 gets line-level balanced XLR outs. Both have a stereo pair of line ins too — often very handy — with adjustable gain. You can attach four pedals: two switches (including a continuous-type damper pedal) and two expression pedals. MIDI connections are via DIN in and out, and a B-type USB socket. In fact MIDI is a strong point here. All controls can generate and respond to controller messages, and (on the YC61 at least) controller numbers are shown in the LCD as you work. There’s also full-blown master keyboard functionality, with up to four independent keyboard zones, working alongside internal sounds or separate from them, and with the option to broadcast bank and patch changes, volume, pan and other information on the recall of a Live Set. Lastly, computers see these keyboards as two-channel in/out audio interfaces. Computer audio can appear at the output jacks (balanced with a menu USB audio volume parameter), but whilst internal sounds can be recorded to a DAW, signals arriving at the input jacks never get into the USB feed. So this is a useful additional feature, but not one that’ll let you record your vocals.

some referencing hardware originals) complement fine amp/speaker simulations, compression, and three varieties of wah, the last of which is easily driven by an expression pedal. There are also digital and analogue-style delays, some respectable reverbs, a lo-fi digital degrader and a resonant low-pass filter which, though not integrated at voice level, is arguably as useful as the EG knob for many jobs. The master multi-effect (which is notably available to the organ section too) is equipped with all the same algorithms except for damper resonance. But it has a couple of its own too: a clean digital-style Tempo Delay, which makes sense of the Tap [tempo] button, and an intriguing Looper Delay. The idea there is to capture riffs (and so on) in real time from any internal sound generator, which you can then solo over using any other. It works in principle, but the maximum loop time of less than 1.5 seconds is just measly, and unlike a dedicated looper there’s no way to define in and out points. It’s a nice idea, but not ready for serious use in this implementation.

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ON TEST YA M A H A C P 8 8 & Y C 6 1

That leaves us with the master level Reverb and EQ. The reverb is a good-sounding fixed hall with about a three‑second decay, and its lack of flexibility would be a problem if it weren’t for the more versatile reverbs available in the multi-effects sections. EQ is a three-band with sweepable mid, working at a truly global level, quite independent of Live Set presets.

CP88 Before I got my hands on the CP88 I’d assumed it to be a Nord Stage competitor. Actually, despite the control panel positively bristling with knobs and switches it’s far closer in scope to a Nord Piano, because it doesn’t have a dedicated organ section. What you do get are dedicated sections for Piano, E Piano and a ‘Sub’ general soundset. Like on the YC61, they can be combined flexibly in split and layer combinations via buttons in each section. I initially questioned the separation of acoustic and electric pianos, because normally you want one or the other, rather than a split or a layer of the two. However, it makes a bit more sense when you spot that each section has its own special repertoire of dedicated effects. Acoustic pianos get dedicated Damper Resonance and a choice of Compression, Distortion, Drive or Chorus. The E Piano section has no fewer than three processors in series: a straightforward Drive, feeding a first multi-effect with various modulation-type effects and wah,

Later Live Both the CP and YC store patches as what’s termed a Live Set. This is basically a snapshot of all parameters, and there’s room for 160 of them. Over half are occupied by factory presets on both models, but all slots are user-writable. Eight dedicated buttons give direct access to pages of eight at a time, or you can use the big clickable encoder next to the LCD to select one from a long list. Loading is pretty much instantaneous. There’s some impressive supporting facilities too. Live Sets can be saved and loaded quickly to and from a USB stick, and individual ones picked out from previously saved bulk dumps. There’s even a Manager page in the menu system for swapping positions and copying them from place to place in the internal memory. When you have them where you want them, a footswitch can be used to advance through the list sequentially.

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The YC61’s drawbars are accompanied by LEDs to show their saved state, which is rather ingenious.

and then a second for Chorus, Flanger and Phaser effects. The Sub section, while we’re at it, has just a single effect of its own, switchable between Chorus/Flanger, Rotary speaker, Tremolo and Distortion. Beyond these insert-type effects there’s a Delay with switchable analogue/ digital characteristics, and a simple Reverb, both accessible to any or all of the sound-generating sections. Finally there’s the same useful three-band Master EQ as on the YC61, but for some reason the CP88 is granted the option of having EQ settings and status stored for individual Live Sets.

Hammer Rite The CP88’s keyboard is one of Yamaha’s own NW-GH (Natural Wood Graded Hammer) models that has wood visible on the sides of the white keys. Under the hand the quite slender and strongly textured black keys made me think more than anything of top-flight acoustic grands from the early part of the 20th century, which is a big plus-point, as far as I’m concerned. There’s no escapement-like resistance on the downstroke, aftertouch or release velocity, but the triple-sensor design makes for reliable retriggering even when keys haven’t been fully released. Octave width is a standard 165mm (which makes a two-keyboard combo with the narrow-span YC61 a subtly strange mismatch), and the white key-dip

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an unremarkable 11mm. There’s a bit of a hollow clatter on the downstroke, much quieter than some cheaper actions, and the release is more muted again. Subjectively, I found the weight and resistance beautifully judged, fast, precise, neither light nor heavy, and musically rewarding for both acoustic and electric piano sounds. Just as I remember the CP4. It’s a good ’un. The piano sounds themselves are identical to, and exhibit the same very high quality as those in the YC61. Here though, fittingly, there are additional grands. The desirable and versatile CFX, C7 and S700 are joined by a plummy Bosendorfer Imperial 290 that reeks of old-world class and a really good, balanced and neutral CFIII. There’s also one more upright, an SU7 with a surprisingly complex and attractive character. You also get the same layered CFXs and CP80s as in the YC61, along with a rather coarse, clangourous (and bizarrely noisy) ‘Digi Piano’. There’s extra value in the E Piano section too, with the YC61’s already generous line up supplemented by two more Rhodes models, a 73 Studio and 74 Stage. These are excellent, amazingly responsive, with pronounced key-release noises. Really, the CP88’s Rhodes sounds are right up there with the very best of the modelling or sample-based competition. Finally the Sub section has 17 pad and string sounds (varying from the

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ON TEST YA M A H A C P 8 8 & Y C 6 1

The CP88’s rear panel is much the same as the YC61’s, but also offers a pair of XLR audio outputs.

Oberheim-esque synthetic to the naturalistic, by way of a single choir and Mellotron strings). Also, 10 organs that encompass a handful of preset Hammond registrations, a single Vox, Farfisa and an (unusual but nice) Elka Panther, plus full-ranks and single-flute church varieties. Then it’s 11 chromatic percussions (think vibraphones, xylophones and some percussive synth bells), and 25 ‘others’ that include basses, synth leads, steel and Strat guitars, a few brass instruments, jazz and Mellotron flutes, and a harmonica. I don’t want to keep repeating myself, but just for the avoidance of doubt, the CP88’s pianos are really good. Timbral complexity across the pitch range, velocity gradation from the NW-GH action, decay phase plausibility and nuanced pedal response all stand up to scrutiny. As with the best of the hardware and sample

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library competition, you’d be hard-pushed to tell you weren’t listening to the real thing on a recording. Only anti-musical, forensic single-note ‘pixel peeping’ would give the game away. Having said that, I was surprised how subtle the piano Damper Resonance effect is: I often struggled to tell whether it was on or off, especially for the CFX. It’s not adjustable like on the YC61. And then, while I’ll readily acknowledge that the provided Sub sounds maintain the quality I saw in the YC61, I struggle to understand why there are so few: fewer than 65 compared to the 300 on the CP4. There’s virtually no naturalistic pop/jazz or orchestral brass, no section or solo woodwind and no orchestral combos, timps or drums. Sound-editing facilities are close to non-existent too. The CP88 doesn’t even get the YC61’s single-knob filter control, so there isn’t a filter at all here, not even lurking amongst the effects. It is perversely equipped with separate envelope Attack and Release knobs, but

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once again they don’t work for all sounds: Attack is too often disabled for some string sounds, for example. I’m also bound to mention the odd misfire in the current soundset (and actually these specific comments hold true for counterparts in the YC61 too). The single glockenspiel sound is out of tune, quite sharp and together with a xylophone it has a big unnecessary baked-in room ambience. The two clavinets, meanwhile, are presented an octave lower than everything else. Another grumble concerns the insert effects, which when toggled on and off cause momentary interruptions to their sound generator’s output. I remember a similar problem with the CP4, and it’s annoying it hasn’t been solved, especially as the same problem is not evident on the YC61. It’s also a shame that the shared Delay effect has no ping-pong option or tap tempo button. On a brighter note, a menu-accessible ‘Advanced Mode’ provides a useful workaround for when the strictly

Alternatives I’ve mentioned Nord’s current line up several times, and those really are the closest competitors in terms of the basic design concept. For piano players the Korg Grandstage and Roland RD-2000 stand out, for different reasons, and organists should check out the ‘new’ Korg-manufactured Vox Continental as well as Roland’s budget-friendly VR-09 and 73-key VR-730.

preconfigured effect provision in each section becomes a limitation. It allows any generator to access sounds outside its normal repertoire: so you could load the Piano section with a bass guitar, say, to access the Compressor there. Sound selection becomes a little less intuitive, because the big categorised selector knobs are disabled, but that’s a small price to pay. It’s also good to see the CP88’s reverb being equipped with a (Decay) Time parameter, adjusting the tail between about 1 and 20 seconds. It’s especially valuable as (unlike the YC61) there aren’t any section-level reverbs.

Back To Black Having spent a lot of time playing and generally getting to know the YC61 and CP88 I really applaud Yamaha for giving the player a more hands-on experience, quite literally. I personally have always clicked with and enjoyed using Nord’s user interfaces, and there is a heck of a lot of overlap here. It feels up to date, in a pleasantly retro sort of way.

I’m also aware that Yamaha have done something clever in dovetailing these models’ capabilities. As I mentioned before, the YC61 bears comparison to a Nord Electro, and the CP88 to a Nord Piano. Each makes for a useful gigging keyboard in its own right of course, and yet put the two together (for only a little more than the cost of a Nord Stage 3, notably) and you have a complementary pairing that solves the age-old problem of playing organ sounds from a hammer action keyboard, and pianos from a waterfall action. Nothing is perfect of course, and the most disappointing aspects for me are how sparse the CP88’s non-piano soundset is, the lost opportunity of the YC61’s transistor organs, and the relative weakness of both keyboards’ abilities as synths. A similar criticism about synth sounds can of course be levelled at the Nord Electro and Piano, which don’t have a pitchbend, mod-wheel or cutoff knob between them. But the Nords score highly in the availability of a large

number of super-vibey vintage string machine, synth, Mellotron/Chamberlin and even Fairlight sounds in their user-configurable sample libraries. These chime with a funky ‘gigging’ keyboard character much better, I think, than Yamaha’s smattering of more generic, general-purpose sounds. I could pick out a few other little negative points too, like how it’s difficult to see panel section divisions in low light, or the sound category knobs, which feel redundant when there are only two sounds in a category, and might have been better employed as endless encoders. Take the YC61 and CP88 on their own terms, though, with their clear strengths in the worlds of tonewheel organs and acoustic/electric pianos respectively, and life with these keyboards could be very sweet. The YC in particular is a hoot to have around: a little ripper! Its Hammond emulation is one of the best out there, all set for two-manual use, and with a beautifully implemented rotary speaker effect that is arguably only beaten by specialist pedals like the Neo Ventilator. The CP88’s grands and Rhodes are uniformly excellent, musically rewarding and fruitfully allied to the much-admired NW-GH action. Construction quality of both models is impressively high, and they look set to endure long, hard working lives. Then there’s the (mostly) glitch-free sound transitions, extensive audio and pedal connections, highly configurable operating systems and clear, useful memory management. This stuff is less sexy on the spec sheet, but often makes all the difference to usability in a whole variety of use contexts. Are they Nord beaters? Not unquestionably, no. But they’re an entirely viable alternative in a hotly contested market sector, with persuasive strengths and characters of their own. ££ CP88 £1729, YC61 £1599. Prices include VAT.

WW www.yamaha.com

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INTERVIEW

Recording Viscerals The rules of heavy metal are well established, but for their latest album, Pigsx7 guitarist and producer Sam Grant added a few of his own.

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WILL STOKES

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t’s probably not a stretch to assume the phrase “don’t let the name put you off” has become something of an epithet to Pigs Pigs Pigs Pigs Pigs Pigs Pigs. In fact, few names could better express the north‑eastern British neo-metal band’s energy, aggression, artistry (it’s seven repetitions, no more, no less) and disarmingly lighthearted self-awareness. The band’s rhythm guitarist and producer Sam Grant cheerily informs me that it’s by no means the most off-kilter band name he’s played under, and since this is a family-friendly magazine I won’t quote him on that any further. The 2018 release of critically acclaimed LP King Of Cowards saw doors open for Pigsx7 that they’d never previously thought possible, considering that up to that point they had thumbed their nose at almost every sensible career move suggested to them. Suffice to say, when third album Viscerals was announced in January 2020 the weight of expectation was considerable. Fortunately, the band couldn’t have been better equipped to achieve what they needed. Not only can they boast a formidable player-producer in the

form of Grant, a true sonic obsessive with a hawk’s eye for detail, they also enjoy on-tap access to a studio lovingly built from the ground up by the man himself in the band’s native Newcastle: Blank Studios.

Pigsx7 rhythm guitarist and producer Sam Grant.

Road Hogs “King Of Cowards had been out around six months,” Sam recalls. “The gigs we were doing were really stepping up with each tour. We were heavily in tour mode, so finding the time to write was a bit more tricky. We booked a couple of weeks at Blank to demo and write, and we already had the recording dates booked in! It was a really daunting prospect, starting the demo process on tracks that we knew in six weeks we’d be recording for an album. That was scary. But we knew what each other had in terms of the formative ideas, so there was a sense of understanding about how it might look and how it would work.” Operating on the boundary of such a delineated genre as metal presented the band with a familiar dichotomy. Sam reflects: “It feels like you have a fork in the road. You could go down one route and say, this band is heavy, ergo they’re a metal band, and so let’s make it sound ‘metal.’ There are established tropes in metal production, with an audience that often expects a particular sound. But then, there’s this sense that when we write and play music together the driving force is about euphoria and catharsis, elevation; and volume and weight are just vehicles for that. We don’t necessarily want to be in a ‘metal band’ per se, but we find ourselves in that space via means. If you want to be in a metal band, then you need to sound like a metal band.”

Make It Sow Central to Sam’s methodology is establishing a finite set of rules to govern the recording process. “It’s an album of rules,” he says. “You set the creative rules and they define your decisions thereafter. The rules going into an album are really important, because in this day and age you can go so far in so many directions that you can lose a sense of direction really quickly. And they’re arbitrary rules — you just set them because they force a creative window.” So what were the rules for Viscerals? Sam rattles them off breezily: “One: the AEA R84 ribbon mic was on every instrument in some form or other. It could be part of a pair, but it has to be there. It acted tonally as a mirror to where I wanted

the album to head. Two: dynamic. That was a big one. It had to have moments of respite. Another one was to find the mid frequencies. Do everything possible to pull mids into the record — overload it! So then there’s a really rich sense of harmonic. Lastly, the plate reverb is the primary reverberance.” Blank’s custom plate reverb was all but installed specifically for Viscerals, and has since become a key element of the studio. Testament to his diligent resourcefulness, Grant built it himself. ‘I’d wanted to do that for a long time, and bought a massive sheet of cold rolled steel a few years ago with the intention of making a plate out of it. I’d decided on King Of Cowards to use lots of room mics, lots of corridor mics. It was natural reverbs. This time around I didn’t want to just get the same sound again. The nice thing was, because I’d built it myself, picked the piezo mics myself [phantom-powered PF5102 JFETs]... Everything about it is its own specific thing. All those things add up to something that’s a distinctive reverberance, one that wouldn’t be on another record. The plate is in the live room, on the wall, so no matter what is going on in there the piezo mics are always plugged in if you want them. You don’t even need to be sending signal to it to get it. It’s a really distinctive sonic signature.

Drum Chops Exacting the right drum sound was foundational to Viscerals, with Sam and engineer John Martindale pulling no punches when it came to ensuring the setup did right by Christopher Morley’s playing.

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INTERVIEW PIGS PIGS PIGS PIGS PIGS PIGS PIGS

Lead guitarist Adam Sykes (left) sets up a Neumann U47 FET, while drummer Christopher Morley adjusts his kit. Sam Grant’s custom-made plate reverb is just visible, hanging on the wall to the left.

“We took a day miking the kit, playing around with different snares, re-heading the drums. We got a nice big Ludwig kit and Chris had some nice Zildjian A-Custom cymbals. It was a felt beater on the kick drum, not a plastic one. A lot of metal these days wants a lot of click and I wanted to try and move away from that clickiness, that brightness. I wanted a darker kit. “The principle was two R84 ribbon mics as overheads. That’s your starting position. Pull up those faders and build a kit around those R84s. There’s a Neumann U87 just above the drummer’s right knee, set to omni, kind of capturing the whole kit. It’s quite an attack-y mic, but it’s a bit of a glue when you bring it in. And then there’s a parallel of that which is crushed through an 1176 compressor, so really distorted. Really thick and fat. “On the kick drum we had a Neumann FET 47 on the outside, that was the primary kick sound, and then inside the kick drum was an AKG D112, lying down. A Sennheiser MD421 was up against the

beater to pull in a bit of attack. With the snare it was more about fatness, so it was the Beyerdynamic M201 which is quite a fat-sounding mic, and that was alongside an AKG C451 pencil mic. They balance nicely; you get a lot of attack and crack off the condenser, and from the M201 you get that more accustomed fat snare sound. The snare was tuned quite low so you get that woof and punch from it, and it was about trying to make sure that was

One of Grant’s self-imposed rules for the recording of Viscerals was that the AEA R84 ribbon microphone be used on every source.

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captured. There was a mic on the bottom of the snare too,” Sam remarks, “but no one cares about that.”

Amping Up With careful adherence to his own set of album-specific recording principles, Sam knew from the off how he planned to capture his own and fellow guitarist Adam Sykes’ performances. “I had a very specific idea of the shape of the guitars before recording them, and that is that the rhythm [guitar] is two takes, panned hard left and hard right so we’ve made that space for the drum kit. That also creates lots of space in the middle for the lead guitar and vocals. One thing about guitars is that tonally they have to be as thick as possible — as harmonically dense as possible. So it was a lot about multiple amps. “I really like, for my guitar in particular, the Orange OR80 and the Vox AC30, totally blown out, and then combining those two amps. With the same take, all you’re doing there is combining the different harmonic deliveries they give to make a thicker tone. Adam uses a Matamp. Really nice-sounding amp. Adam’s sound is a bit more grabby, a bit tighter in the mids, and it sits nicely as a focused centre sound.” As Sam reflects on such aspects of the recording process, small visual cues serve as reminders of his integral role in Pigsx7 as a touring musician, as well as in the studio. His hand-wired OR80 amp head, for instance, is adorned (not by him, I’m told) with a strip of tape labelled ‘Dad’ in commemoration of the birth of his son. Sam continues: “With the guitars it was again the R84 and U87 as a pair, and then

INTERVIEW PIGS PIGS PIGS PIGS PIGS PIGS PIGS

Rhythm guitars were tracked through Grant’s Orange OR80 head.

All the guitars were recorded in stereo, with an AEA R84 and Neuman U87 up close and a pair of Coles 4038s set up further back.

a set of Coles 4038s as room mics and some DPA 4090s in the corridor. I was still always tracking the corridor with the DPAs, across all the instruments, but most of the time they were turned down or off. I’ll put more mics up than I need, knowing that I can just drop them in the mix. “For the bass it was a Fender Bassman head, DI’d out for the most part. It’s a phenomenal head, the Bassman. That was run through a cab with the FET47 and R84 on it as a pair. The bass tone was amazing. The distortion on the Bassman head is so nice compared to a lot of other bass distortions.”

Once More With Squealing Frontman Matt Baty’s powerful vocals by no means represent the only ‘voice’ in Pigsx7; a dynamic that Sam was keen to represent on Viscerals. “Adam, with a lot of his lead stuff, was working with Matt in terms of his lyrics,” he explains. “You know, when one comes forwards the other one’s stepping back, and then when the other goes forwards the first steps back. Then it was a bit of Middle and Sides EQ on it just

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to notch some of the centre frequencies that would otherwise make it difficult to discern the ‘speaking’ parts, so to speak: the solos and the voice, and occasionally the synths.” A Moog Sub Phatty and Korg MS20 were the band’s synths of choice. “Matt can’t track vocals all day because he’d just blow his voice out,” Sam tells me. The answer was to choose their vocal battles carefully and time them wisely, with as few passes as possible. The producer was well aware of the pressure to capitalise on Baty’s performances: “I needed to capture a vibe as much as possible. It was very much a case of the R84, but this time it was running through a BAE 1073. Instead of going through a nice clean ribbon preamp, it was going into a thicker, tonal pre, a much darker one. That was running into the 1176 compressor, because he moves to and from the mic in a performative manner. And then the two Coles mics were a few metres back, going through the ribbon pre. You can over-compress it then, on the 1176, but there’s that bit of space there still. “[Matt’s] vocals have to battle with a really thick, heavy mix. You can go one of two ways about that: you could go with a really bright, edgy U87 or something;

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something that’s going to be detailed and sit on top; or you can go down this route where you’re saying, ‘OK, let’s take on the fight!’ And that’s what we did with this one.” Notwithstanding the unapologetic purity of their modus operandi, it would be remiss of Pigsx7 not to acknowledge the diversity of their growing listener base. Ever shrewd, Sam considers how the band’s potential to extend a sonic hand to first-time listeners and metal uninitiates found its way into the recording process. “Knowing that you’ve got the potential for radio play on an album makes you consider certain aspects. One of the big things with that is how the vocals sit. It’s like, on the one hand radio will want this idea of the lead vocal, but on the flipside the metal world... well, it’s the other end of the spectrum. So you’re playing this game. The easy thing about these mixes, though, is that I don’t actually have to worry too much. Like with the rhythm guitars creating that space, the vocals have a really nice space to sit in the mix. It’s somewhere between modern, contemporary popular music production and Albini-style ‘true to the material’ approach. There’s a simplicity within the

arrangements, within the instrumentation and sonics, which allows for huge amounts of fun to be had in making each sonic as interesting and as deep and rich as possible. And, I guess that’s not normally an attitude that’s taken in rock or metal.” It was in the near-effortless mixing phase that Grant’s attention to detail in the album’s tracking phase paid its dividends. “I’ve got an idea of where it should go,” the producer explains. And the rest of the band are really canny and just hearing the final mix go, ‘Ace!’ Loads of trust. It’s really nice. Because I’m also in the band, playing with the others day in day out, we all know kind of where it is anyway — even just the mood and the direction. I’m not going to miss the point. The mixes tend to happen quite quickly. With my own music, I’ll never get hung up on decisions, I’ll just go on instinct. Sometimes I’ll listen back and change my mind, but it’s nice to have a project which is very quick and pure and direct.”

Blank Canvas Blank Studios’ control room boasts a pair of flush-mounted ATC SCM50ASL monitors: “Super detailed in the mids,” Sam emphasises, “like, super detailed. And when you want to create an album that’s massively stacked in the mids and you want to make it all work, that’s quite important!” At this point, you might be expecting the customary engineer’s ode to a talismanic, 32-channel studio centrepiece. Well, at Blank, there isn’t one. “We haven’t got a console at the studio,” Sam tells me. So, everything is about preamp selection; picking preamps for sources. Over the past 12 years, as a group of engineers, we have had to go from zero. Like, we got five grand off the council in 2008 as part of a business development grant. And that’s been it, really. But what that’s done is forced our hands to understand every bit of equipment that we add to the setup. You end up starting to understand it as a modular tool, the studio. Because we’d never walked into a studio where it’s just like, ‘There you go, there’s

As well as live instruments, Viscerals features the sound of a Korg MS20 and a Moog Sub Phatty.

24 Neve channels,’ or ‘There’s 32 SSL channels...’ That’s not to say that in big studios people don’t have that way of thinking, but I think we might think that way more than most. Because we have to. One of the things for this album in particular was the DACS MicAmp 2. It’s a beautiful preamp for ribbon mics. I sing the praises of the DACS pres. This setting would come to nurture a particularly valuable philosophy for the engineers at Blank, what Sam defines as “working with musicians psychologically, as opposed to with technology.” Much of the sound of Viscerals lies in the very fact that no matter how much of

a clinician he was with his technique and decision-making, Grant’s priority was that Pigsx7’s performance shone unwaveringly throughout the record, above all else. “A performance that’s full of passion and nuance could be recorded on crap gear and sound amazing,” he says. “The primary port of call for us is people. We’re people working with people. I’ve been to a number of studios as a musician and it’s just icy cold! And you just walk away like, ‘Oh, my God. That was hard work.’ It shouldn’t be. It should be such a fun experience. You should almost forget that there’s gear. You should almost forget that there are things between you and the sound.”  

Rather than being based around an analogue console, Blank Studios houses a large collection of outboard preamps (visible in the rack to the lower right of the DAW controller).

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ON TEST

PAUL WHITE

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utput are one of the most adventurous and free-thinking plug-in designers around at the moment. You certainly couldn’t accuse them of relying on digital emulations of rusting tin boxes full of valves and that smell of toasted Bakelite! Their latest creation, Thermal, follows a similar paradigm to their Portal granular processor, which was one of my picks as Gear Of The Year in 2019. But this one’s central purpose is distortion. Just like Portal, there are sophisticated modulation options, additional effects and an X-Y controller, for adjusting parameters assigned to macros. Available for Mac and Windows, it supports the usual AU, VST2/3 and AAX formats, and can be installed on up to four machines. There are plenty of presets, all arranged by the type of sound they’re intended to process, and you can just load one and then fiddle with the X-Y pad to get an idea of how things work. But if you stick to the presets, you’ll miss out!

Parallel Lines Thermal starts by splitting the signal using a three-way filter section. Though it looks like one, this is not a crossover filter, but rather three parallel signal paths, each with its own high- and low-pass filters to restrict the frequency range. The frequency ranges can overlap as much as you like, and can be dragged up/down in level. Each of the three parallel paths, called a Stage, starts with a choice of 19 different analogue and digital distortion types. Post-distortion, each Stage has a choice of nine more conventional effects. Once the three signals are recombined, there are two further effects

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Output Thermal Parallel Distortion Processor

From subtle warmth to full-on modulated mayhem, this plug-in covers a lot of ground.

engines, each of which can again be loaded with any of the nine effect types. The two master effects sections are arranged in series, and are followed by high- and low-pass filtering, a master compressor and a wet/dry mix slider. Fire up Thermal and you’re greeted by a large orange X-Y pad with animated graphics, two macro control knobs, the wet/dry slider and the patch browser. The pad put me in mind of an alien eyeball, floating in a pool of orange lava, but the graphic changes according to settings within the patch. Click on the little ‘controls’ icon, and the window changes to reveal the inner workings of the plug-in. This view is arranged in sections, with the three-channel filter at the top, and below this are sections for the Stage Processing Controls (one tab for each Stage), the two modulation generators, the X-Y pad, effects and master section. The Macro controls are in the panel to the right, and

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clicking on the orange circle above the faders icon takes you back to the simplified plug-in view. Each section has an On/bypass switch and a solo function, and most controls can be linked to a macro. You set this up by dragging the ‘+’ assignment indicator adjacent to each macro knob onto the control you want to include. Do this, and a coloured circle matching the colour of the macro knob will appear next to the control. Dragging up and down on this little circle sets the control range, as indicated by a coloured line segment around the knob in question.

Dirt & Movement Each Stage can be treated with its own distortion type, one effect type (bit reducer, chorus, compressor, stereo delay, filter, flanger, frequency shifter, phaser and reverb), plus stereo width (based on M-S processing) and Tone controls, which work independently of any other filtering in the Stage. There’s also a Refilter tickbox, which is used to filter out unwanted harmonics. The distortion section takes the name of the distortion type currently assigned to it, and is where you choose and adjust the distortion type for the

selected Stage. A dynamic display shows the distortion shape along with the result of any modulations, and the effect can be adjusted using the Drive, Output, Shape, Frequency, Depth and Clip Controls. Controls that aren’t relevant to the current distortion type are helpfully greyed out. Some of the options seem to behave as phase distortions, adding harmonics by changing the shape of the wave, rather than the more conventional approach of clipping or applying soft saturation. Others use digital fold-over, bit reduction, or analogue-style saturation. So, taken as a whole, the options cover a lot of ground but there’s further complexity if you want it: the distortion panel has a feedback section, with adjustable delay time and feedback amount. The two global modulation sections allow just about any parameter to be modulated, and this is where the rhythmic fun really starts. Here, you can draw in complex modulation shapes by clicking and dragging to add and move points. It’s possible to create complex patterns or smooth curves and ramps. Or you can load preset patterns and basic waveforms. These modulators can be tempo sync’ed or left free-running, and

there’s a randomise option if you need some creative inspiration. An adjustable Humanize function applies some random variation to the modulation envelope. Creative use of this section allows complex rhythmic effects to be set up as the ‘+’ symbol can be dragged to other parameter controls to modulate them by any desired amount. Virtually all of Thermal’s controls can be modulated in this way. Both Bipolar and Unipolar modulation characteristics are supported and favourite envelope shapes can be saved for future use.

Feel The Heat Once you have a grasp of the signal flow, working with Thermal is surprisingly intuitive, despite its apparent complexity, and the Stage solo function can make editing settings much easier. The two modulators are the key to adding movement to the

can turn a pad or drone into a useful rhythmic element, but a drum loop can be made to sound more electronic, fatter, edgier or gritty and aggressive. So, do you need Thermal? For anyone with the slightest affinity for experimentation, it is a dream playground. It’s possible to coax some beautiful effects out of Thermal, with treatments ranging from the fairly subtle to the very assertive. Pads can become rhythms, simple drums can become complex, harsh sounds. Warm and polite sounds can be made to pop out of the mix. At one extreme, Thermal can add a new dimension to EDM productions. But it can also sound deliciously smooth in chillout, ambient or cinematic compositions. If you’re a singer-songwriter with just a vocal and piano or acoustic guitar, you’ll get through life without it. But, even then, some of the gentler treatments are great for warming vocals or instruments unobtrusively. Even in mainstream pop Thermal can finesse drum parts and loops, or create interesting backdrops from synth and guitar parts. I’ve only really scratched the surface here: the possibilities are so numerous that I’m still unearthing new uses for Thermal. But already I love it.

“It’s possible to coax some beautiful effects out of Thermal, with treatments ranging from the fairly subtle to the very assertive. ” effects, and a faint visual grid makes it easy to create precise rhythms. Grab a point to move it or grab a line part way along to bend it. Adding a rhythmic modulation to the Drive control of one or more Stages picks out the beat with timbral changes, and this can be further emphasised by using the resonant filter in one of the effects slots and modulating its cutoff frequency. Simple strategies such as these

summary A splendidly anarchic approach to distortion-based processing that could find applications in many musical genres, though it will be of particular appeal to the more experimental musician. I have already found numerous uses for Thermal on my current album project.

The signal path is split into three at the outset and each ‘Stage’ can be filtered, distorted, effected and modulated to your heart’s desire before being recombined and mangled again!

££ £134 including VAT. WW https://output.com

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Managing Sibilance Prevent those pesky esses from ruining your tracks! M AT T H O U G H T O N

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ocal sibilance can not only sound distracting, it can also interact unhelpfully with some of your mix processing. And it can prove a frustratingly stubborn problem. So how best can you stop over-prominent esses compromising the quality of your mix? I’ll take you through some solutions in a moment, but first, let’s consider what makes sibilance problematic. The sound people make when pronouncing the letter ‘s’ can range in character from a full-on lisp, through fairly smooth ess sounds, all the way to a whistle or a ‘sh’, and some of those sounds naturally draw attention more than others. It can be different in pitch, too. Female voices tend to have higher-frequency esses than male ones. All seem to have most energy in the 3-6 kHz region (where our hearing is relatively sensitive), though some have considerable energy higher up as well. Importantly, to make any ess or ‘sh’ sound, we force air through our front teeth, which usually means directing a blast of air directly at the mic. That can be compounded by the mic itself: lots of vocal mics are voiced with a broad boost somewhere around 5kHz to make voices sound closer, breathier or more intimate, and that will boost any sibilance. Furthermore, cheap mics and preamps

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may have relatively low high-frequency headroom, which can lead to distortion, which compounds the problem. Lots of processing and effects that you use in your production can also magnify the sibilant frequencies, and the sibilance can affect the way some processors react. Throw double-tracking and layered backing vocals into the mix, and if the esses in those parts aren’t neatly aligned things can start sounding sloppy, particularly if layered parts are spread across the panorama.

Prevention: Better Than Cure The best time to deal with sibilance is before it’s actually a problem, ie. when you’re recording. An experienced performer should be aware if they naturally emphasise esses more than most, and may or may not have developed mic technique to address that. If not, a cheap, effective trick is to block any gap between their front teeth with dental wax (great if self-recording, though some singers will take more kindly to the suggestion than others!). Alternatively, try the old trick of securing a pencil to the front of the mic with rubber bands. Theory suggests it could cause turbulence near the mic diaphragm, but it doesn’t seem to be a problem in practice and this ‘hack’ can be surprisingly effective. Perhaps a simpler tactic is to hang the mic upside down, from slightly above, with the mic pointing towards the mouth but not

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placed in the path of any sibilant air-blasts. Importantly, this won’t compromise the vocal sound. If none of the above work, the gear could be making an unhelpful contribution. First, check the levels at all points in the chain, and ensure there’s no distortion occurring on loud esses at the mic, preamp or anywhere else. Then consider swapping the mic: if you’re working with the sort of bright-sounding mic described earlier, a different capacitor mic with a relatively flat frequency response might keep you closest to that sound without emphasising the sibilance. Note, however, that EQ boosts you apply later could put you back at square one. A high-quality moving-coil dynamic (such as a Shure SM7b, an Electro‑voice RE20 or a Heil Sound PR-40) might be better, as their construction means they don’t react so dramatically to sibilance, though this means they’ll deliver a slightly changed vocal sound too (which may or may not be a good thing). Another good option is a ribbon mic. These generally sound a little ‘darker’ than capacitors due to their HF roll off, but they can usually take substantial HF EQ boosts without losing that smoothness (probably because the ribbon’s resonance is very low, well outside the wanted audio range). Modern preamps, even those built into affordable interfaces, aren’t usually a problem, but cheap analogue gear

Here, I’ve used Steinberg’s SpectraLayers (rather crudely!) to highlight some ess sounds recorded on a cheap mic. These sounds dominate the upper (blue) area, and the darker lines mean there’s more energy at those frequencies. The main energy here is around 4-6 kHz, but you can see plenty of information above 10kHz too.

that distorts easily can interact in a nasty way with essing. So do watch those levels and listen out for problems, and if you’re deliberately driving a preamp into saturation, keep an ear out for the impact of that on those esses.

Conventional De-essing Whatever you do when recording, there will inevitably be occasions when you need to address sibilance later, whether because a part was poorly recorded or because sibilance is emphasised as a side-effect of mix processing and effects. Usually, you’d want to address problem sibilance at the beginning of the signal chain, though as I’ll explain there are other places de-essing might be required. In most cases, the first thing I try is a simple, static EQ cut. This can often tackle sibilance very well, but can sometimes unhelpfully colour the vocal sound more generally, potentially compromising the intelligibility of some lyrics too, so listen carefully for such unwanted side-effects. This tactic can be very effective where the essing has become offensive only because you’ve EQ’ed the part for another reason. For example, when applying a high-shelf boost to brighten a vocal, an accompanying dip in the sibilance region won’t generally detract from the overall sense of ‘lift’. If

you’re struggling to identify the offending frequencies — the best results can sometimes require two or three narrow cuts — try looping playback on an ess sound and use a frequency analyser plug-in to guide you. Melda’s free MAnalyzer, for example, allows you to zoom in and will display the centre frequency of any peaks. If you slow down its response, it’s easier still to see where the energy is building up. When EQ can’t reduce the sibilance without unacceptable collateral damage, it’s time to consider more sophisticated de-essing tools. Most de-esser plug-ins work in real time as the audio is fed through them, though there are some offline tools too which I’ll discuss later. Most work in broadly the same way: they’re essentially compressors with a side-chain filter that you tune so they act only where there’s sibilance. They may be ‘full band’, and duck the whole signal like a regular compressor, or ‘split-band’ and duck only the sound above a crossover frequency. They may or may not allow user control over various other parameters but, usually, I don’t find that they actually need to be hugely tweakable. Something that I think is important is the ability to monitor the side-chain/detector frequencies and, ideally, the delta signal (so you hear only what’s being removed, which is helpful to understand the side-effects). This makes setup quick and easy, which is the whole point of this sort of de-esser. You might sometimes find it tricky to arrive at one setting that does it all, and in that case you might try stacking two

It may look a little DIY, but the pencil-on-a-mic trick can be very effective at taming sibilance.

de-essers in series, each contributing to the overall gain reduction, or perhaps try automating the threshold. Multiband compressors can be configured as de-essers too, of course, and sometimes they’ll offer you a little more useful control over things like the attack and release time. Dynamic EQs are similar tools, in that they’re also threshold-based dynamics processors, but you get finer control over which frequencies are pulled down and by how much. Some can be configured to operate separately on the Left, Right, Mid or Sides channels, which can be useful if you need to reduce sibilance on a part with more than one singer, or where there are vocals and instruments (or bleed from

Don’t write off simple EQ moves as a remedy, particularly if the sibilance only becomes an audible problem after applying a high-shelf boost.

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instruments recorded using other mics) on the same recording. It’s also worth bearing in mind that all of these types of dynamics processors can be set to react to an external side-chain signal. Probably the most accurate de-essing processors are offline ones, such as the one in iZotope’s RX. Both Melodyne 5 and Revoice Pro are also able to identify esses and other unpitched sounds, and these allow you to drag these sounds down in level as you go through to do other edits. This can be a great way to tackle recorded sibilance in a mix prep stage (when you might do any denoising and tweak clip

FabFilter’s Pro-DS gives you loads of control, but particularly handy is the headphone icon, which allows you to hear only what the de-esser is removing — a great help when fine-tuning things to minimise unwanted side-effects.

envelopes, fades and so on) but, even if your DAW supports the ARA plug-in format, I find they can be a bit trickier to fit into your workflow once you’ve already started mixing.

Chop & Change Talking of ‘mix prep stages’, probably my favourite option is to use my DAW’s editing tools to manually chop out and attenuate problem esses. I know it sounds laborious, but it has its advantages, and with experience things like esses and breaths become really easy to spot in a waveform — easier still if you have a DAW with

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a spectrogram view (eg. Reaper). So if you already go through and prep the various audio files before you start mixing, it’s not a great leap further to cut the esses out and put them on another DAW track. This allows you to ride/automate the sibilance channel fader or a low-pass filter to soften the esses. It prevents the esses changing in level as you add processing to your main vocal part, and they won’t accidentally trigger any threshold-dependent processors you use on the main vocal track either. (Just remember that if you want to include all the esses in any other vocal processing, you’ll need to route both these tracks to a bus and apply that processing there.) You’ll also have a ready-made solution to another problem I sometimes encounter: a stray ess that just sounds a bit weird; an ess that’s perhaps more whistly or lispy than others and draws attention. All you need to do is mute/chop out the bad ess, and copy and paste a good one in its place.

Effects & Processing I mentioned earlier that sibilance can interact with your processors and effects. A typical mix includes all sorts of processing, most of which will come downstream of de-essing, which is usually done early in the signal chain. You’ll often find that you need to tweak your de-essing settings in light of channel EQ and compression, distortion, saturation or harmonic enhancement processing. That said, there are other places you can make useful adjustments. For example, you can EQ the side-chain of a compressor to change how it reacts to sibilance. Send effects such as reverb and delay can draw

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If you already prep your files in other ways, it’s not a huge amount of extra effort to cut esses out to another track where you can change their level, apply low-pass filtering and other processing, to give you much greater control over any recorded sibilance.

unwanted attention to esses, but in these cases I find it’s better to use filtering or a de-esser on the aux track, just before the effects plug-in. That way, you don’t have to change the sound of your ‘dry’ vocal, but only the sound entering the effect. Finally, also listen out for the effects of any bus processing. I tend to use the ‘top-down’ approach to EQ, and often apply fairly assertive HF boosts on the master bus. These naturally bring up any sibilance, but rather than EQ the sibilance away, it’s often easier just to route the vocals around that master bus EQ.

How Much Is Enough? It’s important that the cure isn’t worse than the disease. All the close attention you pay to sibilance while treating it can cause you to lose perspective. Listening fatigue can cause you to be too light-handed, because your ears grow accustomed to the sibilance. But on the other hand, the intensity of your focus on sibilance can cause you to be overzealous in treating it, resulting in a part that sounds somewhat dark and lispy. So how do you know if you’re striking the right balance? As always, regular breaks will help you avoid listening fatigue, as will avoiding playing things back on loop for too long while you tweak. But it’s also a good idea to have some reference material to hand. Many years ago, Mike Senior recommended Natalie Imbruglia’s ‘Torn’ for this: he found the essing is just that bit too much, so it served to tell him when he needed to pull it down a bit. I’ve been happily using that tip ever since. Whatever you use as a reference for sibilance, it’s good to have one to help you retain perspective!

D I S C O V E R

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ON TEST

Plugin Alliance

Brainworx Amek EQ200 EQ Plug-in This Brainworx EQ is inspired by esoteric mastering hardware. M AT T H O U G H T O N

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espite the Amek branding, Plugin Alliance’s new Brainworx Amek EQ200 is actually inspired by owner Dirk Ulrich’s love of certain mastering hardware, notably EQs made by Sontec and GML. Before you rush out to compare it with those frighteningly expensive mastering units (or plug-in emulations of them), I should point out that this doesn’t directly emulate either. Rather, the Brainworx team set out to create a hybrid that builds on what Ulrich considers the most desirable control elements and tonal characteristics of those processors. They’ve taken the opportunity to extend the functionality in software too, with new features and some that were introduced on previous Plugin Alliance software. The Amek EQ200 is available in

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the common plug-in formats for Mac and Windows. Installation on my MacBook Pro (Mac OS 10.14.1) was hassle free, and I tested the AU and VST2 and 3 versions in Reaper and Cubase.

Workflow This is a dual-channel five‑band parametric EQ, each band of which is individually bypassable, but it boasts many other features. There are additional 12dB/octave high- and low-pass filters plus a Mono-maker elliptical filter, a Mid-Sides‑based stereo width control, and a THD harmonic distortion control. There’s generous overlap between the five parametric bands. The two channels can be linked/unlinked as you prefer, and can be configured to process the Left and Right or Mid and Sides channels. So there’s plenty of flexibility here, though you can’t select L-R/M-S per band, which would be nice. There’s no frequency analyser or visual representation of the changes you make to the frequency response, of the kind we’ve grown accustomed to seeing on so many digital EQs, but I actually like that. Much as I like to think I’m not swayed by what I see, I find I make different decisions, and largely better ones, with less assertive boosts

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and cuts, when I use an EQ with this sort of interface — where the sound and the position of the knobs are all I have to go on — than I do when using something like, say, FabFilter Pro-Q3. Despite the profusion of virtual knobs, buttons and LEDs, the layout is clean and intuitive, particularly when you have the lower panel hidden. The LP/HP filters are at the outside of each channel, and on each channel the bands run, reading left to right, from low to high. Gain is at the top,

Plugin Alliance Brainworx Amek EQ200 $399 pros • Sounds great. • Useful additional mastering facilities. • Gain-scaling, invert and auto-listen features are genuinely useful.

cons • Expensive to buy outright.

summary The Amek EQ200 sounds good, is thoughtfully designed and encourages you to use your ears.

bandwidth in the middle and frequency at the bottom. Beneath each gain control is a switch to change the scale from ±15 to ±7 dB. The latter generally proved more useful to me in mastering and on master-bus EQ — both roles in which this EQ serves well — as it makes fine control much easier. Worth noting is that if you’ve already moved the knob from its neutral position, and then click the ±7/±15 button, you’ll change the amount of boost/cut on that band. I’d have preferred it if the value remained constant (when still in range) but the knob moved accordingly, and a global button for this might be helpful too. Nonetheless, it’s a useful facility. Between the two channels, are more controls. The Gain Scale knob and Invert button are thoughtful features. Many is the time that I found I’d boosted/cut the right frequencies, but had generally overcooked things, and backing things off on the Gain Scale control delivered the desired result really quickly. On occasion, I also found it useful to exaggerate what I’d done — rather like a magnifying glass that led me to refine some decisions before dialling the Gain Scale back again. Its real use, though, is for boosting to find undesirable resonant frequencies, then hitting Invert to take them out, and then scaling those cuts to taste. Very handy. The EQ In button is a bypass, as you’d expect, except that it applies not only to the EQ bands but also the various other top and bottom panel controls. By default the two channels are linked for L-R stereo operation, so clicking either bypass button bypasses both channels. In the lower panel, though, you can choose to unlink the channel controls, including the bypass. (Very occasionally, in v1.0, changing the link/unlink setting triggered a bug in the master output level control, making it apply only to one channel.) What you can’t do is to first unlink the channels and apply a corrective offset to one, and then apply a linked boost/cut to the same band. If, say, I’d applied a 1dB boost to the Sides channel while unlinked, and then I linked the channels and tweaked the same band in the Mid, I’d lose that 1dB Sides boost; the Sides would jump to and follow the Mid setting. I’d prefer it if there were an option for the second action to apply an offset to the first, so boosting the Mid by 1dB would increase my 1dB

Sides boost to 2dB. Of course, this being software you could use two instances to achieve the same thing. The remaining control in the middle of the GUI engages an Auto Listen facility. Switch this on, and when you click and drag a Q or frequency control it will solo that band. More helpful still is that you can achieve the same thing using a modifier key. There are also modifier keys for fine control and to jump between the default and last settings — the actual keys for these modifiers vary according to the plug-in format and the OS, as detailed in the PDF manual. A feature in the Advanced panel is Plug-in Alliance’s Tolerance Modelling Technology. The idea is to simulate the per-channel differences of analogue hardware that result from variations in the

The top panel includes a number of useful utilities, not least a 32-step undo. An ABCD facility allows you to compare different settings and to copy and paste between them, without having to save/load presets. It’s the sort of unremarkable, workmanlike feature that I wish every developer included. A Reset button sensibly ‘zeros’ the controls on the current ABCD instance, rather than for the whole preset (for the latter, you’d use a zero’d preset). Nearby, you’ll find useful Mid and Sides solo buttons, a GUI scaling button, and another to show/hide the lower panel. Finally, there’s V-Gain, which models the noise inherent in analogue electronics and gives you control over the level of that noise floor. This defaults to being on, and can become quite obvious once you’ve lifted the high end and applied some compression and limiting. I can’t complain that it’s there (some folk like this sort of thing, and Brainworx are not alone in modelling this stuff) but an early move for me was to switch it off and save that over the default settings preset!

“The top panel includes a number of useful utilities, not least a 32-step undo.” tolerances of components. You can choose one of two stereo modes. The first has both channels identical, while the second uses different profiles for each. In either mode, you can select which two of up to 20 modelled profiles are used. I can’t say I found it of much use in this particular plug-in — it’s a feature that really comes into its own when using many instances on different channels, such as you might with PA’s SSL channel-strip emulation. But it’s a nice extra, and if you work with lots of top-down EQ on the stereo mix bus and various subgroups it could be useful here. This lower panel also hosts the master input/output level controls, each with a 0 to -60 dB LED-style meter. A correlation and stereo balance meter nestles between the Mono Maker and Stereo Width controls. A Mid-Sides stereo-mode selector and THD control complete the lower-panel selection. The former is self-explanatory, the THD control less so. It threw me a little at first because the ‘T‘ stands for ‘third’, not ‘total’ as I’d anticipated. Turn the knob clockwise and you increase the prominence of the third harmonic in the modelled distortion, and when you then boost an EQ band, its associated harmonics rise faster than the overall frequency response. It’s all pretty subtle stuff in the grand scheme of things, but the character is generally easy on the ear, and this is one thing that sets this EQ apart from a lot of others I’ve used.

Sound Judgment Whether the Amek EQ200 sounds precisely like the mastering hardware that inspired it I can’t really say; I don‘t have easy access to those units and it isn’t intended as an accurate model anyway. But I did compare it with some of Acustica ’s ‘sampled’ mastering EQs, including their Green 3, which is based on the GML8200. I compared them both on a range of material for mastering and bus processing. The Amek EQ200 generally acquitted itself fairly well in these comparisons, though sounded a touch ‘softer’ in the high end in particular. The Amek EQ200 won hands-down when it came to versatility, though, given the additional band and the various lower-panel features. The price of the Amek EQ200 isn’t trivial but might make one of Plugin Alliance’s subscription plans very appealing. If you already subscribe to their Mix & Master or Mega bundles, you just got a very fine plug-in for ‘free’ and I’d urge you to check it out. ££ $399. Also included in subscription bundles from $14.99 per month.

WW www.plugin-alliance.com

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The audio interface is at the centre of every modern studio, but how do you choose the right one for you? SAM INGLIS

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hirty years ago, control rooms contained big mixers and racks of audio processing equipment. Today, the computer is the cold robot heart of the studio, and software has replaced the console and outboard. This change means that one piece of equipment has become indispensable. You can’t run a computer-based studio without an audio interface — but how do you know which interface is right for your computer-based studio?

Analogue Inputs Getting sound into a computer is a two-stage process. First, an analogue signal is converted to a stream of numbers. Then this stream of numbers is fed into the computer. This second stage is the core function of an audio interface, but nearly all of them do both. In other words, most interfaces can accept analogue signals directly. What’s more, they can often take two different types of analogue input. A ‘line’ input expects signals at a standard studio level such as is generated by synthesizers, mixers or mic preamps. However, we also want to plug microphones and electric guitars directly into our audio interfaces. These put out feebler and less predictable signals, which have to be preamplified before they can be digitised. Most project-studio interfaces thus feature analogue inputs with mic preamps and/or sockets to plug a guitar into. Often, you’ll find dual-purpose sockets which combine an XLR for mic input and a quarter-inch jack for line input. So one of the most basic questions to ask yourself is: how many analogue inputs do I need, and of what type? As a first

Many interfaces use space-saving ‘combi’ input sockets that can accept either an XLR cable from a microphone, or a line input on quarter-inch jack.

step towards answering this question, count up the Preamp gain is most commonly adjusted number of sources you’re using an analogue potentiometer. going to want to connect simultaneously. If you only ever record yourself, you might only need one or two inputs, even if you’re building complex tracks by overdubbing. If you record bands live, you may require enough inputs to cover a multi‑miked drum kit, several other instruments and vocals. Some interfaces allow preamp gain to be controlled digitally, In a studio where the audio which offers increased precision as well as recallability. interface is the only item of hardware, you’ll need it to have as many allows you to leave mics and line-level mic preamps as you’re ever likely to want sources permanently connected and to connect mics. But if you plan to use switch them on the fly as needed, your audio interface with a hardware rather than having to re-plug. mixer, or you only ever record synths, or • On some interfaces, mic preamp you have other equipment already that gain is adjusted digitally. This is more can amplify the signals from your mics, precise than using an analogue control you might well prefer to get an interface and means that settings can be fully that has only line-level I/O. recalled and sometimes even stored with your DAW project. Breaking A Tie • If you use capacitor microphones, This only gets us so far, because there you’ll need your interface to offer are a lot of audio interfaces that seem to phantom power. Nearly all do so, offer identical analogue input facilities. but sometimes this is only switchable If you have to choose between several globally or in groups. This can be models with the same features, you relevant if you want to connect things should also consider the following: like ribbon mics, which shouldn’t encounter phantom. • The audio performance of interfaces Analogue Outputs varies, and some of these differences can be important. Read the ‘Which Audio interfaces perform a comparable Specs Matter?’ box for more details. two-stage job at the ‘back end’ of the • Some interfaces have separate mic and system, spitting digital audio out of the line sockets for the same inputs. This computer and turning it into an analogue

Nearly all audio interfaces have at least one built-in headphone output, and many have two. Often these show up in your DAW software as separate outputs, but not always.

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TECHNIQUE HOW TO CHOOSE AN AUDIO INTERFACE

signal. The most basic reason for this is so that we can connect speakers or headphones to hear sound coming out of it! Nearly all interfaces therefore feature at least one pair of line-level outputs and at least one stereo headphone socket. Again, the key question to answer is: how many outputs are you likely to need? Bear in mind that there are almost no audio interfaces with more than two headphone outputs. For recording more than two musicians at once, you’ll thus need a dedicated multichannel headphone amp. This, in turn, will need to be fed from its own pair of line-level interface outputs. Bear in mind, too, that unless you have an outboard monitor controller with built-in speaker switching, you’ll also need a separate pair of line-level outputs for each pair of speakers you have. Remember, also, that outputs aren’t only used to feed monitoring systems. To integrate hardware compressors or equalisers into your software mixes, you’ll need line-level outputs to feed them and line-level inputs to receive the processed signal. The same goes if you want to mix on an analogue console.

Going Deeper Once again, considering only the raw numbers will probably leave you considering lots of interfaces with the same basic output arrangement. If so, it’s time to ask yourself some more detailed questions that will help you find the most suitable interface for your needs.

Monitor control features on audio interfaces vary considerably. If your interface doesn’t offer features such as speaker A/B switching, talkback and mono, you may need to budget for an additional hardware monitor controller.

• M  ost audio interfaces provide some monitor control, but the features on offer vary wildly. At its most basic, this is a simple level control for one pair of outputs. At its most sophisticated, you might have configurable control over multiple outputs, along with additional features such as talkback, dim, speaker switching, a button for checking your mixes in mono, and so on. • Headphone outputs don’t necessarily all appear as separate destinations in your recording software. Sometimes they duplicate what’s feeding one pair of line outputs, though you’ll often have some choice about which pair. • If you use modular synthesizers such as Eurorack devices, you might want to consider buying an audio interface with ‘DC-coupled’ outputs that can generate a steady-state or DC voltage. These can produce the control voltage signals used to drive modular synths, allowing you to sequence your synths in software. (See the ‘Modular Interfacing’ article

elsewhere in this issue for an in-depth guide to this.)

Room For Expansion A little research into larger interfaces will reveal that once you get beyond eight mic inputs, your options narrow dramatically. Yet eight mic inputs is not enough to track a band. What if you need 12, 16 or even more mic inputs? One option is a mixer that incorporates an audio interface. Products like the Zoom L-12 and L-20, Behringer’s X series, the Tascam Model 16, Soundcraft’s UI mixers, PreSonus’s StudioLive consoles and the QSC TouchMix‑30 Pro provide lots of inputs, and can take care of live sound and cue mixing as well as recording. They can be excellent choices for people who work primarily with bands, but there’s a lot of variation in the features on offer. A detailed comparison is outside the scope of this article — but very much within the scope of Chris Korff’s piece elsewhere in this issue!

Which Protocol? Four main connection protocols are in widespread use for connecting audio interfaces to Macs and PCs, though Ethernet is less relevant to project studios at present. I’ve summed up the main pros and cons of each in this table.

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Connector

Pros

Cons

USB

Affordable and simple to use, future-proof and available on all computers

Low-latency performance can be indifferent, interfaces generally can’t be used in multiples

Thunderbolt

Usually offers very fast low-latency performance, easy to use, often permits interfaces to be used in multiples

Not universal on Windows computers, more expensive than USB, cables are costly

PCIe

Generally offers the best low-latency performance, often permits a choice of converters

Not available on laptops, typically a high-end professional option

Ethernet

Extremely flexible, especially in multiroom installations, allows very long cable runs, multiple computers can share access to system

Complex and can be hard to set up, low latency requires a dedicated Ethernet card in the computer, rival incompatible standards

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Some manufacturers offer very similar interfaces with both Thunderbolt and USB connectivity.

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TECHNIQUE HOW TO CHOOSE AN AUDIO INTERFACE

The two main formats used for digital I/O on small interfaces are coaxial, which can carry stereo audio in the S/PDIF format, and optical, which can carry either stereo S/PDIF or eight-channel ADAT Lightpipe signals.

A second option is to attach more than one audio interface to your computer. Usually, this means buying two interfaces from the same manufacturer, and even then, multiple working is hardly ever supported on USB audio interfaces. Interfaces that can be connected in multiples usually do so using Thunderbolt ports or PCIe slots; not every computer has these, and interfaces that exploit them tend to be more expensive than USB equivalents.

Digital Inputs Fortunately, there’s a third option: adding more analogue inputs to an existing interface. Interfaces often provide more paths to the computer than they have analogue inputs. These paths are fed by digital inputs, which can accept signals already converted to digital data by some other device. Indeed, some interfaces only offer digital inputs, the idea being that users can assemble a bespoke system by choosing their converters separately.

rackmount units on the market that offer eight channels of analogue-to-digital conversion with ADAT connections. Beyond the project-studio world, you’ll find many other digital formats, most of which need not concern us here. Assuming your expansion plans are centred around ADAT and/or S/PDIF, there are a few considerations to bear in mind:

Digital audio can be encoded electrically, but it can also be represented optically, using pulses of light. The key point is that the encoding scheme and • Optical connectors are usually the physical medium are independent of switchable between ADAT and S/PDIF, one another. The most common optical but this is not always the case, so do connector is thus used for two different, check that optical S/PDIF is supported incompatible types of digital audio data if you need it. — one of which can also be sent over an • Many interfaces carry two pairs of electrical wire! That data type is S/PDIF. optical connectors. Sometimes these Connecting an optical or electrical S/PDIF cable from the output of one device to the input of another will carry two channels of digital audio between them. However, the same cables that carry optical S/PDIF signals are also used for a multichannel format known as ADAT or Lightpipe. If you’re working at the standard 44.1 or 48 kHz sample rates, a single Lightpipe connection can carry eight channels of digital audio in one direction. In the project-studio world, this Most Windows music software uses the ASIO (Audio Streaming makes it ideal for affordably Input Output) protocol to communicate with the audio interface. expanding an audio interface, As this is a third-party standard, you’ll always need to install and there are lots of a driver before you can use your interface.

Which Specs Matter? All manufacturers publish technical data about their products. Unfortunately, this often leaves out important measurements such as low-latency performance, but specifications can still be useful in deciding which model is right for us. In some respects, the technical performance of modern audio interfaces is so good as to be a non-issue. For instance, all of them have a flat frequency response throughout the audible range, so won’t audibly change the timbre of sounds going in or out. Many also offer a dynamic range of at least 110dB on both inputs and outputs, which is far more than is needed to capture any real-world signal. So an even higher dynamic range figure, for example, is perhaps best treated as a sign that the manufacturer knows what they’re doing, rather than something that will directly benefit our recordings. Specifications that make a real difference include gain range and maximum input level for mic preamps. The larger the gain range, the wider the range of input signal levels that can be

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accommodated. The maximum input level gives you a reference point for that versatility. If this is high — say, +23dBu — you’ll know that you can safely record drums and other loud sources without fear of clipping. But unless the gain range is also high, it might be a struggle to get a respectable signal level on quiet sources, such as speech recorded with a dynamic mic. There is quite a lot of variation between interfaces, so it’s worth thinking about what applications really matter to you. (Gain range is sometimes defined using maximum and minimum values, in which case you can calculate the range by subtracting the minimum from the maximum. If, for example, the minimum gain is -5dB and the maximum is +55dB, the total gain range is 60dB.) If you record quiet sources, you want to be able to do so without adding unwanted noise. All mic preamps introduce some noise into the signal path, but some perform better than others in this respect. The key measurement here is Equivalent Input Noise or EIN. Look for the largest

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negative number and be aware that A-weighted figures look better than unweighted ones. The very best preamps manage about -129dB unweighted, which equates to about -132dB A-weighted. On the output side, maximum output level for line outputs can be important if you want to connect your interface to old-school studio hardware. Most professional outboard is aligned for a maximum level of +20 or +24 dBu, but not all interfaces can generate this. This can mean that the outboard won’t deliver optimum performance, and in extreme cases, you might struggle to get a hardware compressor to do anything if you can’t feed it a strong enough level from your interface! The built-in headphone outputs on audio interfaces also vary, and some can drive headphones louder than others. Unfortunately, this is a specification that is often presented in different ways or not at all, making it quite hard to compare rival products.

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TECHNIQUE HOW TO CHOOSE AN AUDIO INTERFACE

Larger interfaces often include sophisticated DSP mixing features, which are controlled from dedicated Mac OS and Windows programs.

can be used to add two eight-channel ADAT expanders, for a total of 16 extra channels. In other cases, though, the limit is fixed at eight channels, and the second connector is used only at 88.2 or 96 kHz sample rates; at these rates, twice as much data needs to be moved around, and eight channels’ worth of audio will no longer ‘fit’ down a single optical link. • Not all interfaces allow you to use ADAT and

Small interfaces sometimes implement direct monitoring using a simple balance control, which allows some of the input signal to be directly fed into your headphones and monitors.

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S/PDIF expansion at the same time, even if both sockets are available. • In order to communicate successfully, digital devices need a shared timing reference, so once you introduce any sort of digital connection into your studio, you’ll need to learn about clocking. When only two devices are involved, this is usually straightforward, and it will be explained in the product manuals.

Latency & Software In an ideal world, we’d monitor what’s being recorded through our recording software. The problem is that getting the data into and out of the computer takes time, and if it takes too much time, there’s an audible lag between playing a note and hearing it on headphones or speakers. The total time taken for a signal to travel through a recording system, from source to monitor system, is known as the round-trip latency. Some people are more sensitive to latency than others, but once it gets much above 10ms, most will notice it. Interfaces are supposed to report their latency to the host computer, but

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many do not do so accurately. Latency is adjusted using a setting called buffer size. The lower the buffer size, the lower the latency — and the greater the demand on the computer’s Central Processing Unit. For any given buffer size, some interfaces will perform better than others, both in terms of the CPU load and of the actual latency they deliver. Thunderbolt and PCIe interfaces often outperform USB interfaces here, but another important factor is the driver software that handles data transfer between interface and computer. Many USB interfaces are ‘class compliant’, and can use the Apple driver built into the Mac OS operating system. This is good enough for most purposes, but interfaces that employ custom driver software usually perform even better. This includes some USB interfaces, and all Thunderbolt and PCIe models. On Windows, recording software uses the ASIO driver format developed by Steinberg. This isn’t part of Windows, so you’ll always need to install a driver, and the quality of these is quite variable. Many manufacturers of USB interfaces license third-party driver software from developers such as Thesycon, whilst other manufacturers code their own drivers. The latter usually offer better performance, but the situation is complex and it isn’t always easy to tell what driver a given

If you don’t have other rackmount equipment, you may well find that a desktop interface is more convenient.

interface uses. For more detail and for rigorous measurements of low-latency performance on Windows computers, a visit to Vin Curigliano’s DAWbench website is essential.

Mixing In In general, low-latency performance is better today than 10 years ago. But even with the best drivers, a round-trip latency of under 5ms can be hard to achieve, especially on USB interfaces. For this reason, many audio interfaces build in a mixer which allows us to hear input signals without waiting for them to pass through the computer and recording software. On some small ‘desktop’ interfaces, this mixer is controlled using a simple knob that adjusts a balance between input signal and playback from your recording software. Where more than a couple of inputs and outputs are concerned, though, manufacturers build in a digital mixer controlled from software. Manufacturers take varied approaches to the design of digital mixers and the software that controls them. Some build in very powerful and complex mixers with endless routing options. Others concentrate on simplicity and ease of use, offering just enough functionality to cater for typical use cases. Yet others build in not only mixing features but also plug-in equalisers, compressors, reverbs and other signal processors. Digital mixers in interfaces can sometimes be controlled remotely from tablets and phones, and even sometimes within recording software by the same manufacturer. Which of these approaches suits you is a matter of personal taste, but be aware that all of them can be implemented well or badly, and it pays to do some research. Read SOS reviews and check user forums online before parting with your cash. This is an aspect of interface design that’s easily overlooked, but it will affect your day-to-day experience with the product like nothing else.

Other Factors Besides the core features I’ve already described, there are other features that

vary between audio interfaces. If you’re still struggling to choose, perhaps these considerations will help to tip the balance: • S  mall interfaces can often be ‘bus powered’ through the USB or Thunderbolt cable, and some offer no alternative. Bus powering

• L evel meters provide indispensable information about the amplitude of signals going in and out of your interface. This information is always available in software, but most interfaces also provide hardware meters too. If the reassurance and immediacy of good hardware metering is important to you, be aware that the

Clear, comprehensive metering can be a big plus for some users.

is convenient and makes the interface more portable, but can limit performance. For example, bus-powered interfaces often cannot drive headphones as loud as mains-powered rivals. • Some mains-powered interfaces require an external power supply. This can be inconvenient, and a lost or damaged PSU will put your studio out of action until it can be replaced. • Larger interfaces almost always adhere to the 19-inch rackmount format, but smaller ones come in many shapes and sizes, so ergonomic differences might affect your decision. If you are going to use your interface in a rack, do you want all the sockets on the back?

functionality on offer is very variable, ranging from a single LED on some interfaces to highly configurable, colour touchscreen displays on others. • Some interfaces include MIDI In and Out ports for digital control of synthesizers. Finally, always remember that an audio interface requires committed support from the manufacturer, for instance by providing driver updates. Some manufacturers have a better track record than others when it comes to providing this support, especially for discontinued models. Choose right, and your interface should last you through many OS and computer upgrades!

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F E AT U R E

Many of the features that won live sound engineers over to digital desks also have huge potential in the studio. We check out some of the best options.

PreSonus StudioLive

CHRIS KORFF

W

hen large-format digital consoles from the likes of Sony, Tascam, SSL and AMS Neve first appeared, they were targeted at recording studios as well as broadcast and post-production. They never came to dominate the recording sector in the same way, though, and as the notion of

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‘analogue warmth’ gained traction around the turn of the century, they started to become rather unfashionable. At about the same time, digital technology began to take over the live-sound market, as live productions became ever more ambitious and required ever higher channel counts. Advanced features like remote control, audio networking, scene saving and recall,

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personal monitor mixing, huge amounts of DSP, and direct-to-USB recording started trickling down into the lower end of the live-sound market, and it wasn’t long before studio shut-ins saw their potential for recording applications. In particular, the fact that so many of them can also work as multichannel audio interfaces means that modern digital mixers can fulfil a number of roles: preamplification,

A-D/D-A conversion, routing and mixing are built into one box. So, whether you need a fader-packed console for hands-on mixing, you do enough live work to justify owning a digital desk, or you’re simply drawn to the idea of having an interface and mixing engine in one unit, let’s see what the current crop of digital mixers can offer you.

Behringer X Series

PreSonus StudioLive PreSonus’ original StudioLive console was specifically designed for both live and recording applications, combining familiar analogue-like ‘one channel per fader’ operation while offering 32 channels of computer recording over Firewire. The range has grown steadily since the first one appeared in 2003, adopting the USB protocol along the way, and now spans everything from large-format 64-channel consoles to the dinky 1U‑rack StudioLive 16R. StudioLive mixers also offer an impressive amount of integration with PreSonus’ Studio One DAW. For example, you can use a StudioLive mixer as a Studio One DAW controller, and because both console and DAW use the same Fat Channel signal

processing, you can even transfer mixer settings between them.

Behringer X Series The X32 had a profound effect on the live music scene, and arguably did more than any other mixer to convert the lower echelons of live sound over to digital. For the price of a decent analogue board, small venues and gigging bands could enjoy flying faders, virtual soundchecks, LCD scribble strips, remote stageboxes,

personal monitor mixers, and more. Like the StudioLive range, the X32 series has expanded to cover a number of different form factors, like the compact Producer edition and various rackmounting options. More recently, Behringer have also launched the intriguing Wing mixer, which also looks to have recording applications.

Soundcraft UI24R As live engineers started patrolling their venues with iPads, some bright spark

Soundcraft UI24R

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F E AT U R E D I G I TA L M I X E R S I N T H E S T U D I O

Allen & Heath Qu-SB

started to wonder whether mixers needed a control surface at all. And then someone (possibly the same person) went further, and asked if we couldn’t just hide all the mixer stuff inside a stagebox, chuck a WiFi router in there for good measure, and banish stage snakes altogether... The Soundcraft UI series was born. The UI12 and UI16 offer excellent value for live situations where faders aren’t needed and space is at a premium, but the flagship UI24R is the one to look at if you’re looking for a recording mixer-cum-interface. Like most of the other mixers here, it can operate as an audio interface and a direct-to-disk recorder, but the UI24R is slightly unusual in that it can do both at the same time, offering a convenient redundant setup for when failure is not an option.

including the Qu-Pac and Qu-SB. Both are based on the tech from their larger Qu-series consoles, and they offer a similar amount of analogue I/O, but differ chiefly in terms of the amount of physical control they offer. The Qu-SB can only be controlled remotely, via Ethernet or Wi-Fi, whereas the Qu-Pac sports

Allen & Heath Qu-Pac & Qu-SB Soundcraft aren’t the only proponents of the stagebox-format mixer: Allen & Heath also make some enticing options,

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Mackie DL32R

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a touchscreen and encoder, allowing you to access all of the main functions directly. Naturally, both can also function as multichannel audio interfaces over USB.

Mackie DL Series Despite being nearly a decade old, the Mackie DL1608 remains the mixer of

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F E AT U R E D I G I TA L M I X E R S I N T H E S T U D I O

Zoom LiveTrak L-12

choice for our Executive Editor Paul White, mostly because the remote Master Fader app is so intuitive and quick to use. You can now get various rackmounting DL-series mixers, of which the flagship DL32R, with its 32 analogue inputs and 14 assignable mix outs, would make a formidable studio centrepiece. And if faders are a must-have, Mackie also offer the DC16, a substantial hardware controller with 17 faders and encoders, extra I/O via Dante, and a talkback section.

Zoom LiveTrak More than ‘just’ a digital mixer, Zoom’s LiveTrak devices are also highly capable stand-alone multitrackers — so while most of the other models here can make ‘rough and ready’ multitrack recordings of all their inputs without a computer, the LiveTraks can do overdubs and punch-ins, have a proper project management

system, facilitate song position markers, and so on, all in addition to being able to work as audio interfaces. They also all boast multiple headphone outs with individual foldback mixes, and the baby of the range — the LiveTrak L-8 — can even run on batteries.

Yamaha TF-Rack One company who never gave up on the idea of digital studio mixers is Yamaha, whose venerable 01 and DM series mixers are still in production today — indeed, our Technical Editor Hugh Robjohns still has and uses a DM1000 in his home studio.

QSC TouchMix-30 Pro

They also make the (large and expensive) Nuage post-production consoles and, more realistically for most of us, the TF series of live-sound desks. With its intuitive touchscreen, rackmount form factor and optional Dante connectivity, the TF-Rack looks like a particularly studio-friendly option.

QSC TouchMix-30 Pro

Yamaha TF-Rack

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QSC’s TouchMix series are highly regarded by live engineers for their sound quality, robustness and usability. Although they weren’t originally conceived with studio recording in mind, a major firmware update to the flagship in the range, the TouchMix-30 Pro, added 32 channels of bi-directional audio interfacing via USB. It also has a number of unusual but very handy extra features, including DCA and mute groups, a real-time analyser, and a graphic EQ room-tuning wizard.  

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TECHNIQUE

CV & The Audio Interface Everything you need to know to get your DAW talking to your modular system via your audio interface. ROBIN VINCENT

T

he roads of MIDI between computer and hardware synthesizers are well-trodden, but once you become a devotee to control voltage (CV) you need different pathways to find your way from DAW to Eurorack and back again. Let’s have a look at what you’ll need to pack in your backpack for the journey, along with your sandwiches and a nice flask of tea.

The Theory Audio signals and CV signals are all variations in electrical potential, and they run along the same sorts of cable. Part of the point of modular synthesis is that there is some crossover between the content of audio and CV signals. An LFO, for example, generates a cycling control voltage that we use to modulate parameters within a synthesizer. As you increase the rate or

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speed of the LFO, it reaches a frequency that’s within the range of human hearing, whereupon we can treat it as an oscillator generating an audio signal. Conversely, an audio signal can be used as a modulation source, and many audio oscillators can be slowed down to a frequency below the threshold of hearing. The outputs on computer audio interfaces are, as you’d expect, designed to output audio signals. But since audio and control voltage signals are so similar, surely we can also use our DAWs to generate CV signals and our audio interface to pipe these into our modular synths? Well, sometimes. The issue here is that although audio and CV share the same method of delivery, their content can be completely different. Audio voltages are constantly changing, or otherwise we wouldn’t hear the results as audio. Crucially, however, a CV can be a static, unvarying value

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— a DC voltage or, if you like, a signal with a frequency of zero Hertz. In a system designed to record and play back audio signals, the ability to reproduce signals below 5Hz or so is a mixed blessing. The DAW software itself generally has no problem with very low frequencies, but many audio interfaces are deliberately designed to filter them out; they can’t be heard, and they can cause interference or DC offsets that could distort our audio or damage our speakers. And so the biggest obstacle to using our DAW to record, play back or generate CV is an audio interface that filters out DC and very lowfrequency signals.

DC-coupling There are two ways of dealing with this. One is a bit of a fudge, while the other requires that you have the right sort of audio interface. Most audio interfaces have

what is called AC-coupled outputs, where a capacitor is used to filter out the extreme low frequencies. This is a disaster for CV, as it cannot output the sort of slow-moving or static values we need. Outputs without this filter are described as DC-coupled, and are increasingly common in audio interfaces as the use of CV is on the rise. These are capable of generating stable DC voltages and very slowly changing control values, which is exactly what we need to drive a modular synth. One manufacturer whose interfaces have always been DC-coupled are MOTU, while DC-coupled outputs and inputs (but not necessarily both) can also be found on interfaces by RME, Universal Audio, NI, Apogee and PreSonus, who’ve now made it a feature across their entire range. However, just because an interface has DC-coupled outputs doesn’t necessarily mean it’ll be a fully capable source of CV. The Eurorack format specifies a control voltage range of ±5V, but many portable and bus-powered interfaces are not capable of such wide voltage swings. For example, I recently reviewed in this magazine three

USB-powered audio interfaces: the RME Babyface Pro FS, Native Instruments Komplete Audio 6 and MOTU M4. The KA6 could output a range of ±2V, the M4 achieved about ±3V whereas the Babyface Pro could produce the whole ±5V. A good way to get a rough idea of an interface’s capabilities in this department is to check the specs to find out the maximum output level on the line outs: NI quote +11.5dBu for the KA6, MOTU +16dBu for the M4 and RME +19dBu for the Babyface Pro.

AC-coupling If you have an AC-coupled audio interface then you are mostly out of luck — except, as I say, that there is a possible fudge. This involves using amplitude modulation to sneak the low frequencies through the interface by superimposing high-frequency signals on them and then removing the AC component afterwards. This requires a filter, which is a relatively simple circuit to build. Expert Sleepers used to sell a little module that did it, but now they’ve turned it into an algorithm within their Disting module, which does the same job. Going the other way,

you’ll need something that will convert your CV into an amplitude-modulated signal that you can sneak past the AC-coupled audio inputs. The Disting again has an algorithm that’ll do this, and I’ve not come across anything else that does. Provided the software you’re using (which we’ll come onto in a moment) supports an AC mode, then, you can either make a few circuits or buy a number of Disting modules and be all set for sending CV signals from your DAW to your Eurorack and back again. Realistically, though, it’s probably much simpler just to get yourself a DC-coupled audio interface.

CV Generation In DAWs There are several bits of music software that can generate and output control voltage. There’s CV Toolkit from Spektro Audio, MOTU Volta, Reaktor Blocks, Dialog Audio SQ4 and Audulus to name but a few. But for this article, I’m going to highlight Bitwig Studio, Ableton Live CV Tools and Silent Way from Expert Sleepers. Bitwig has a couple of devices within its modular structure that can generate and

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TECHNIQUE MODUL AR INTERFACING

Expert Sleeper’s Disting module allows you to partake of the dark art of AC-coupling.

receive CV. HW CV Out is a simple knob that sends voltage out of a specified audio output. You can then start applying internal modulators like LFOs, step sequencers, randomisers or even MPE expressions to that knob to produce the sort of CV output you’re after. You can create fabulously complex forms of modulation within The Grid environment and attach it to that knob to send it to your modular. The other device is the HW CV Instrument which gives a pitch output and a gate output to provide a melodious signal for sequencing or playing from a keyboard. The octave range of the intended oscillator has to be tuned and calibrated to match the pitch to the voltage. This is done by routing the output of the oscillator back into Bitwig and running a calibration process. This can be a bit fiddly; you need to get input levels right and use quite a bit of trial and error to get it to scan correctly. But, once accomplished, you can send tunes out from Bitwig into that particular oscillator. If you want to try another oscillator then you’ll have to recalibrate. If you want to go in the other direction of controlling instruments or devices within Bitwig from your modular you can add an HW CV In modulator to any device, specify

the input, and then direct the voltage to whatever parameter you wish. It’s really very straightforward. Bitwig CV devices have an AC mode and so can generate amplitude-modulated CV signals and interpret them coming in. In theory, then, Bitwig should work well with regular AC-coupled audio interfaces provided you have that bunch of Distings to hand. In practice, however, I had great

There’s also a CV Shaper in which you can create any wave shape you like. CV Utility is more like Bitwig’s voltage knob: you can apply all sorts of modulation to it from internal devices or MIDI controllers attached to Live. All of these rely on you having a DC-coupled audio interface, as there’s no AC mode. For CV coming into Live there’s a CV In device that takes any incoming voltage and lets you map it to any software parameter. This has a Pitch mode that can be used with an AC-coupled interface to convert an oscillator pitch into CV in real time, so that’s at least something. With their ES-1 modules, Disting and DC-coupled Eurorack audio module, Expert Sleepers are undoubtedly the experts in CV and DAW integration. Silent Way is their bundle of CV tools that you can use as plug-ins in any DAW, and offers a comprehensive range of functions. Silent Way is also available as a Rack Extension for Reason and as virtual modules for VCV Rack opening up both of those modular environments to copious amounts of CV control. There’s a ton of stuff in here for generating voltages as pitch or modulation, gates and triggers, quantising, controllers and followers. Amongst them is the Voice Controller, which does the same Pitch and Gate thing as the devices in Bitwig and

“There’s definitely something in the way CV works that offers something worth pursuing.”

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difficulty getting the calibration of the Bitwig CV instrument to work in AC mode, whereas I had no trouble with the Silent Way plug-ins.

Live CV Tools Ableton Live version 10.1 introduced the CV Tools pack of devices for Max For Live, designed to handle the CV side of things. It has a CV Instrument device very much like the one in Bitwig for handling melodic output of pitch and gate signals. A CV Triggers device works like a drum machine, and a Rotating Rhythm Generator can generate all sorts of interesting patterns. For modulation there’s a simple CV LFO which can output various waveforms.

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Live, but adds a whole bunch of envelopes, LFOs, portamento and detuning. It creates an entire synthesizer voice in voltage which you could connect to all sorts of bits of your modular. Silent Way also has an AC Encoder which will do all the hard work in using amplitude modulation to get through an AC-coupled audio interface.

Making It Work So, armed with the right software and the right audio interface, is it easy to get your DAW talking to your modular? Yes, for the most part. The easiest way to get CV flowing into and out of your computer is via the Expert Sleepers ES-8 or ES-9 DC-coupled audio interface modules. They connect via USB and become the audio engine for your DAW. The ES-8 has four inputs and eight outputs and is expandable via ADAT, whereas the ES-9 has 14 inputs, eight outputs plus ADAT and S/PDIF, the only difficulty being that there are no microphone, instrument or line inputs such as you might need for recording non-modular signals into your DAW. The ES-9 does at least provide audio outputs

for monitoring non-CV audio. Otherwise, DC-coupled audio interfaces like those from MOTU and PreSonus can do the job really well. There’s one caveat, which is that if your DC-coupled audio interface has balanced TRS outputs, you’ll need what’s called a ‘floating-ring’ cable. If you try to use a standard mono-to-mono patch cable from your interface to your modular, you’ll only get the positive half of the signal. To get the full range you need a TRS-to-TS cable that’s wired as a floating ring rather than being simply a stereo-to-mono cable. Expert Sleepers can supply them, and you can also use insert cables that have TRS on one end going to two TS jacks. This has the added bonus of giving you positive CV on one jack and an inverted version on the other. So, to summarise, there are two possible signal paths for driving a modular from your computer: • C  omputer > CV-capable software > DC‑coupled audio interface > floating‑ring cable > modular. • Computer > CV-capable software with

AC mode > AC-coupled audio interface > Disting > modular.

The Digital Alternative Whether it’s worth the effort to have control voltage running through your digital workspace is really up to you. The alternative is to look at MIDI-to-CV and CV‑to-MIDI options that tend to work right out of the box. You could, for instance, have a Hexinverter Mutant Brain MIDI-to-CV converter module, which will convert a single MIDI connection into 16 Eurorack‑compatible outputs from notes to modulation, gates, triggers and clocks. Going the other way, you could use the Befaco VCMC to take CV notes, gates and modulations and map them to MIDI notes, events and controllers. MIDI-to-CV tends to be unipolar, though, and doesn’t have the feel of CV control. It feels like you’re sequencing MIDI equipment rather than experimenting with the impact, interaction and mixing of voltage. There’s definitely something in the way CV works that offers something worth pursuing.

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w w w . s o u n d o n s o u n d . c owww.scvdistribution.co.uk m / September 2020

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Arturia

AudioFuse

Studio

USB Audio Interface Arturia’s audio interfaces just keep getting better. SAM INGLIS

W

hen it appeared in 2017, the Arturia AudioFuse was probably the most versatile desktop audio interface ever made. Not only did it combine mic and line-level audio I/O with digital inputs in S/PDIF and ADAT format, it also offered a blizzard of additional features. These included monitor control, talkback and a built-in USB hub, along with re-amping capability, insert points on both analogue ins, phono inputs and an earth terminal that could be used for connecting a turntable, mini-jack MIDI In

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and Out and both full-sized and mini-jack headphone sockets. The AudioFuse also sounded very good, thanks in part to Arturia’s class-leading DiscretePRO mic preamp design. This cornucopia of features was, however, accompanied by a few quirks. Some of these have been ironed out in firmware updates, but I never grew to love the AudioFuse’s unusual hybrid direct monitoring system, its rather crowded layout, or its dependence on generic audio drivers. And, with my reviewer’s hat on, I could never quite figure out what sort of user this incredibly comprehensive feature set was aimed at.

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The AudioFuse remains a current product, and was joined last year by the AudioFuse 8Pre. The 8Pre showcases Arturia’s mic preamps in a much more streamlined interface targeted squarely at bands, drummers and other people needing to record multitrack audio. By ditching some of the original AudioFuse’s more unconventional features, simplifying others and repackaging the whole thing in a spacious and smart 1U format, Arturia really allowed its good qualities to flourish, and the 8Pre seems to me an extremely strong product.

A Longer Fuse This year finally sees the AudioFuse relinquish its crown as the most versatile desktop interface ever made, but only because Arturia have made one that’s even more versatile. The AudioFuse Studio replicates all of the AudioFuse’s

The AudioFuse Studio crams a huge amount of connectivity on to its back panel, without being so bafflingly crowded as the original.

many and various I/O options, adds some new ones of its own, and packages the sum in a larger desktop case that is, to my mind, both better looking and better laid-out. If the original was the Swiss Army knife of audio interfaces, its new sibling is an entire central European cutlery drawer. The original AudioFuse was designed to be bus-powered, but in practice this isn’t possible on some computers and requires some compromises in performance. Sensibly enough, Arturia haven’t tried to make it an option on the Studio, which is powered instead from an external PSU with a locking adaptor. The Studio also doesn’t have the AudioFuse’s rigid lid to protect it on the road, presumably reflecting the fact that you’re less likely to be using it on roads. However, permanent powering means the AudioFuse Studio can be used as a standalone mixer/preamp/ format converter/multi-bladed implement without a computer attached, which is

probably more useful. When you do attach a Mac or PC, you do so through a USB 3 Type-C socket, but the Studio is fully compatible with USB 2 and comes with both types of cable. As on the 8Pre, Arturia have binned off the AudioFuse’s direct monitoring system, which is no bad thing in my view. What you get instead is a more conventional internal digital cue mixer, controlled from the AudioFuse Control Center utility. Where this differs from the 8Pre’s version is that there are three separate mix panels. Labelled Main, Cue 1 and Cue 2, each of these can draw on all the available analogue and digital inputs. This addresses one of the 8Pre’s few shortcomings, allowing you to send different cue mixes to the two headphone outs and to the main outputs.

One of many very nice touches is that the signal feeding Headphones B — which can be any of these three sources — can also be made to feed the second pair of speaker outputs, with its level governed by Headphones B’s level control. So not only could you choose to feed a different mix to your alternate speaker pair, if desired, but you also gain independent volume control for the main and secondary speakers. (That’s in addition to the already thoughtful option to apply a fixed volume offset to the second speaker pair...)

Front To Back The AudioFuse Studio has four identical front-panel inputs on combi XLR/jack sockets. Like those on the 8Pre, each can operate as a mic input or a high-impedance input, and connecting a line-level source

Arturia AudioFuse Studio £799 pros • Amazingly comprehensive feature set. • Excellent sound quality. • More flexible and ergonomically better than the original AudioFuse.

cons • With so many features on offer, it’s probably inevitable that you’ll end up paying for some you never use. • Generic drivers offer adequate rather than stellar low-latency performance.

summary Creating an interface that’s even more feature-rich than the original AudioFuse was a challenge, but Arturia have done it. More importantly, they’ve also ironed out some of its quirks, and the Studio is a pleasure to use.

The AudioFuse Control Center utility provides access to three separate cue-mixing panels. Each can ‘see’ all of the AudioFuse Studio’s inputs, although it’s also possible to hide I/O you’re not using.

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ON TEST ARTURIA AUDIOFUSE STUDIO

The Audio Settings dialogue is appropriately exhaustive.

bypasses the mic preamp for the cleanest possible signal. Each also has its own dedicated insert point on the rear panel. What’s new compared with other AudioFuses is that each channel now has a latching/momentary Listen button in addition to its phantom power, pad, polarity and source selection buttons. Depending on how you have things configured in the Control Center, this either triggers the Solo function for that channel in the cue mixer, or sends the input to a separate PFL bus. The PFL bus bypasses the channel faders in the mixer, and can be routed to any or all of the speakers or either headphone output. I think this is a great addition, allowing the anxious engineer to check up on any suspect sources in his or her mix during a take, without affecting what the musicians are hearing. As well as the four front-panel inputs, there are four further analogue input paths, all of which do double duty. Each has a quarter-inch, line-level balanced jack, but if you use the turntable inputs, those override inputs 5/6, while 7/8 share a two-lane computer highway with yet another input type: Bluetooth. Pressing the button at the top right of the main panel causes the Studio to proffer itself for pairing, and I had no trouble playing music into it from my phone. Bluetooth input seems to blithely trample upon everything that’s going on in the Control Center mixers, though. The Studio’s three pairs of quarter-inch output jacks are as versatile as you’d expect, especially the ‘auxiliary’ outputs.

Bundled Software Most audio interfaces these days come with software to sweeten the deal, and Arturia’s offering in this department is particularly toothsome. Analog Lab Lite is a virtual instrument which collects together some choice presets from their V Collection, but you also get Arturia’s 3 Preamps You’ll Actually Use, plus their Minifilter, Comp FET-76, Delay Tape-201 and Reverb Plate-140. In each case, you’re getting the full version of the plug-in rather than a taster course, and if you were to buy all of these separately it’d cost you a significant chunk of the price of the AudioFuse itself.

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Both of these can be switched into a re-amping mode, allowing you to feed a DI’d guitar recording out to an amplifier at the correct level and with the appropriate source impedance. Routing the DI’d input directly to an auxiliary output during tracking allows you to record both direct and miked sounds, with minimal monitoring latency. This output pair is also DC-coupled, meaning they can be used to generate control voltages for use with a modular synth. Compared with the AudioFuse, there’s no new digital I/O as such on the Studio, but a second pair of optical sockets allows the full eight-channel ADAT count to be maintained at 88.2 and 96 kHz.

Light The Fuse As I’ve already mentioned, the original AudioFuse is extraordinarily comprehensive, and the Studio version even more so. On paper, though, the differences perhaps don’t seem as great as all that. Isn’t the Studio version just a double-width AudioFuse with an extra pair of mic/line/instrument inputs and a pair of auxiliary outputs? And given that this makes it much less portable, is it really worth spending the extra to get the Studio? To my mind, it certainly is worth the difference, but that’s not so much

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because of the relatively modest expansion in I/O. It’s because the Studio is much more pleasant to work with. The form factor is a big improvement, the new cue-mixing and direct monitoring arrangements are excellent and the PFL option is really handy. Along with numerous smaller enhancements, it all adds up to a feeling of refinement and ergonomic design that was slightly lacking in the original — while, of course, the Studio version retains that device’s excellent sound quality. Arturia are still offering generic drivers with the AudioFuse range, rather than coding their own. These will never quite match the performance offered by rivals such as RME, MOTU or Focusrite who use custom drivers, and that’s a shame, but at least the Studio’s improved direct monitoring features mean you’re less likely to need to run it at super-low buffer sizes. I liked the AudioFuse Studio a lot, and although I didn’t get the chance to test them together, I think that adding an AudioFuse 8Pre as an ADAT expander would give you a killer setup for many band and location recording duties. ££ £799 including VAT. WW www.sourcedistribution.co.uk WW www.arturia.com

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ON TEST

ROBIN VINCENT

W

hen Novation revealed the Launchkey Mini MkIII last year you knew it wouldn’t be long before the rest of the Launchkey range followed suit. It’s taken longer than expected, but everything that made the Mini MkIII a very cool little controller has been stretched out into the full-sized and infinitely more playable Launchkey MkIII.

What’s New? The model under review is the largest in the range at 61 keys. Along with the usual 25- and 49-key options there’s now a 37-key version for those people who find the 49 too big or the 25 too small. The features are the same across the range except that only the 49 and 61 have the fader bank. This new version looks sedate and serious. It’s all clean lines, angles and elegance and it does that thing where it instantly makes the very capable MkII look garish and unfashionable. What were they thinking with the coloured underbelly of the MkI and MkII? It was really easy to unpack, slid out of the box with one hand and I was delighted to see foam packed under the keys.

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Novation Launchkey MkIII Controller Keyboard

The third generation Launchkeys offer versatile control over hardware and software alike. It’s pleasingly compact for a 61-note controller at only 26cm deep, which is a bonus for our crowded desktops and studio spaces. Beyond the sharp lines the most obvious change is that the pads and faders have swapped positions. Pads are now over on the left with faders in the middle, just off-centre, and only the transport controls on the right. I wonder how much customer feedback went into agonising over that decision? The hardware has a poise to it that raises both a smile and your expectations. It’s still made of plastic but doesn’t seem as clunky as many 61-keyed MIDI controllers can feel. The look is cemented

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by the replacement of the red three-digit display with a cool blue-backed LCD with two rows of 16 characters. The pads are nicely set just that little bit lower and all the other buttons follow the style and height. The knobs are upright, cylindrical, feel slightly rubberised to the touch, and are taller and much more solid than the ones on the Launchkey Mini. They give a good, smooth resistance. The faders have a similar feel and are now accompanied by RGB buttons which will no doubt make themselves useful. The keys themselves have a relatively light, synthy feel. The white keys are shiny and grip your fingers whereas the black keys have a less grippy matt

texture. They are velocity-sensitive, offering three selectable curves or can be turned off. There’s no aftertouch on the keys but the pads are both velocity- and pressure-sensitive and will do single or polyphonic aftertouch. Rounding off the physicality we have proper rubberised pitch and modulation wheels, and out the back there’s a single USB socket, sustain pedal input, and an old-fashioned and very welcome 5-pin DIN MIDI output. It’s USB powered and that’s the only cable that comes in the box. This brings me to one of my only complaints about the Launckey MkIII: Most computers keep the power on their USB ports even when turned off, so the only way to stop the glow of those pads is to pull the USB cable out. Before we plug it into Ableton Live (for which it is largely designed) there are a number of useful standalone features baked into the controller that don’t require a computer at all. There’s a wonderfully fun Arpeggiator, eight scales and inbuilt chords that you can use with any external MIDI gear via the MIDI Out on the back.

Arpeggiator The Arpeggiator comes from the Launchkey Mini and has all the usual directional and timing features you’d find with any arp, but it also has some special generative sauce in the flavour of Mutate and Deviate. Mutate takes the notes you are holding and starts to add octaves, harmonics, intervals and finally chaos, depending on how far you dial it in. Deviate does the same with rests dropping in all kinds of rhythms before hitting complete randomness. They transform a familiar function into something enormously fun and creative — it can keep you entertained for hours. The main functions of Type, Rate, Octave and Rhythm are mapped out on the keyboard and accessible by holding the Shift key. The first five knobs take on the roles of Tempo, Swing, Gate Length, Mutate and Deviate respectively and are again accessible by holding the Shift key. One of my biggest gripes about the Launchpad Mini is that you had to hold a button to manipulate the Arp, which prevented you from playing and manipulating at the same time. Well, it looks like Novation have taken that on board because you can now lock the Arp controls on by briefly holding down the button. The functions

A five-pin MIDI Out and a host of onboard features mean the Launchkeys play especially well with hardware synths.

from the keyboard are now selected on the pads meaning that you can keep playing with one hand while accessing the Arp functions with the other. That’s a huge improvement.

Screen This might be a good point to talk about another improvement, which is the screen. Every time you move a knob or touch a pad the screen tells you what you’ve just done. This sounds a bit after-the-fact but it’s a terribly useful confirmation that helps make the workflow more intuitive. It works especially well with the usage of the pads in Arp mode and selecting Scales and Chords where you are faced with a cryptic row of coloured pads. Tap a pad and the screen tells you its current function and you find yourself a lot less baffled.

Scales/Chords There are eight scales to choose from, which quantise the notes on the keyboard. Then, using the Chord mode, the pads can become instant scale-appropriate chord generators. You get a row of triads with inversions, a row of 7ths, 9ths and 6/9ths which you can reach via the up/ down arrows. If you go into User Chord mode you can create and save your own bank of chords. Just hold a pad and play up to six notes to enter your chord. It’s at this point that you’re wondering where you’re going to get an extra finger to play a six-note chord with one hand, but you don’t need to play a chord, just play the notes you want to appear in the chord one after another. You can transpose the chord with the up/down arrows and even copy the chord to the next pad to create a bit of a progression. Finally we have Fixed Chord mode where you hold the button, play

a chord and then you can play that same chord up and down the keyboard transposing aswe go. With or without a computer the Launchkey MkIII has a lot to offer as a standalone MIDI controller. All the knobs, pads and sliders send out MIDI and are mappable to software or hardware MIDI devices. In Drum Mode the pads are set to MIDI note numbers which are set by default to trigger GM drum sounds. But it also potentially gives you another place to play or trigger things from a separate MIDI channel to the keyboard. There’s no split function on this controller, but that’s a sneaky way of getting some dual control action.

Ableton The Novation integration and workflow with Ableton Live is mature and well documented. Working with this

Novation Launchkey MkIII From £159 pros • Style and design. • Ableton Live integration. • Not just for Live. • Arpeggiator Mutate and Deviate. • MIDI Out port. • Can store for custom modes.

cons • No Off button. • No aftertouch. • Transport buttons don’t light up.

summary Novation bring both style and substance with the MkIII Launchkey, offering near-perfect Ableton Live integration along with some tasty standalone features and a fun generative arpeggiator.

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ON TEST N O V AT I O N L A U N C H K E Y M K I I I

combination has become fast, intuitive and almost idiot-proof. Provided you are running Live 10.1.15 or above then everything is done for you and the Launchkey puts itself into Session Mode ready to give you control over pretty much everything. The two rows of pads give you a colour‑matched display of two rows of Session clips. You trigger them by tapping the pad or you can launch a session by pressing the big side arrow button. But that’s only for the top row. The second row can double as a Stop, Solo or Mute for the clip above it. You can navigate around the Session using the up/down arrows and the Track left/right buttons. Ableton Live displays a red rectangle around the 16 clips currently in view on the pads. The only slight annoyance here is that to move the view one step to the right requires tapping the Track button eight times to move the track selection all the way over to track nine. You can get around this by using the buttons under the faders to select the far right or left track first, but you’d think there’d be an easier way just to move the view window left a bit or right a bit. The Pots take on panning by default but you can quickly employ them as volume, sends or Device controls with the touch of Shift+Pad. You can switch the pots to Pickup Mode in the settings which means that when switching between different types of control the position of the pots are saved and are only reactivated when you turn the pot back through that saved value. This prevents sudden controller changes when you first turn a knob. On the 49- and 61-key versions the faders take on volume control over the current eight tracks in view but these can also be switched to control sends or Device parameters. However, it won’t let you select what the Pots are currently set to and vice versa. The process of recording into clips, mixing, manipulating and moving on is really smooth. You can work and play on a track while hardly touching the mouse at all. A couple of additional buttons in the MkIII make this easier. ‘Capture MIDI’ is the coolest new feature in Live that the MkIII has a dedicated button for and it brilliantly stuffs whatever you were playing into the nearest clip. But you’ve also got a Quantise button, Click On/ Off and my most commonly executed command: undo.

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Custom Modes Novation look after the configuration of their hardware devices using a browser-based piece of software called Components, in which you can set up the MIDI functionality of every knob, button, fader and pad. You can create custom modes that give the faders the right MIDI CC# to match your particular hardware synth or turn

Device selection is now also a breeze: press the Device Select button and then tap the appropriate pad. Under that button is Device Lock which locks the controls to the currently selected device so you can keep on tweaking while wandering off to other devices or other tracks. One last feature is found with the button with three dots. This makes a bunch of pads emulate the arrow keys on a keyboard and the enter key. You can now quickly navigate around all sorts of menus and parameters and select them. It’s also a very neat way of zipping around the session clips to choose one to fire outside of the 2x8 pad window. There are things you can’t do like create tracks, browse and load plug-ins or samples, so the integration and functionality is more about playing within a session you’ve built. So if you were to load up a drum kit and a bunch of instruments you could go to town with the Launchkey, building up entire sessions filled with clips. And within that preloaded environment it’s so easy to jam away with yourself without ever having to leave the keyboard.

Conclusion Throughout writing this review I’ve been referring back to the images and

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pads into program changes. You then upload that Custom Mode to your controller. The brilliant thing about the MkIII is that you can now store four Custom Modes independently for the Pots, Faders and Pads in the Launchkey itself. You can leave the computer behind and swap modes whenever you need to.

manual of the MkII and I can’t quite believe how old and clunky it appears. In comparison the MkIII is a work of art in keys, RGB pads and plastic. The sense of style and design is spot on and it looks totally fabulous on my desk. It feels like Novation have worked hard to pull in a more fluid Ableton Live integration that means you can stay at the keyboard, stay being creative, rather than dropping to the mouse. That said, it’s the MIDI Out socket and those standalone features that elevate this device from being an Ableton Live-based keyboard to being a controller that’s creatively useful in any MIDI situation. It’s no longer tied to your computer and can take on our hardware needs too. It’s also worth mentioning that the Launchkey MkIII has scripts for enhanced functionality with both Reason and Logic, and there’s also good old HUI support for all the other DAWs and you can get busy mapping the controls to whatever you want. So while it’s a superb choice for Ableton Live users, the Launchkey has more than enough to keep things interesting for users of anything. ££ Launchkey 25 MkIII £159.99, 37 £179.99, 49 £209.99, 61 £259.99. Prices include VAT. WW www.novationmusic.com

Rostam Batmanglij & Danielle Haim

Secrets Of The Mix Engineers: Ariel Rechtshaid & Rostam Batmanglij Haim’s latest album was made with a mix of old-school and modern techniques and equipment. PAUL TINGEN

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aim’s third album, Women In Music Pt. III, was not only a UK number one, but also exceptionally well received in other ways. The album scored a whopping 89 out of 100 on Metacritic, with writers noting its experimental and eclectic nature, and saw the band moving away from the sunny Californian pop/rock of their first two records. Women In Music Pt. III was produced by the band’s singer, guitarist and drummer Danielle Haim, together with her boyfriend and producer Ariel Rechtshaid, and Rostam

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Batmanglij. The latter was once both the producer for, and a member of, Vampire Weekend (see the SOS May 2010, www. soundonsound.com/people/secrets-mix-e ngineers-justin-gerrish-rostam-batmanglij), but since 2016 has ventured out on his own, both as a solo artist and a producer for such artists as Frank Ocean, Lykke Li, Wet and Maggie Rogers, among others. Talking via Skype, Rechtshaid and Batmanglij explain that much of the sound of Women In Music Pt. III was the result of recording and treating real instruments to make them sound old, and then sampling them to make them sound modern again.

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“Yeah, it is sort of what we did,” comments Batmanglij, laughing. “In general many samples are taken from old recordings, so we first made things sound like old recordings, and then we did things to make them sound gritty and grainy so they sound like a sample, and then the final step was to make them sound big! Some of our choices were really us flying by the seats of our pants, as we were trying to retrace the steps of how old recordings were made, and then making that sound like a sample. That was a huge goal. “We didn’t actually sit down and say, ‘We are going to make something sound old and then make it sound new,’ clarifies Rechtshaid. “We were not really consciously talking that way. But speaking for myself, I am very influenced by hip-hop production, and coming up as a kid in LA in the ‘90s I got my hands on a sampler and tried to find the same breakbeat they used on, for example, an Ice Cube record. I have always been a student of recording history, and I also am a huge fan of music software. We tried to combine these two, but that’s obviously not going to sound retro. We were pushing plug-ins and our laptops as far they could go,

Photo: Grant Spanier

INSIDE TRACK

Ariel Rechtshaid at his Burbank studio in 2017.

Photo: Ashley Beliveau

‘I’ve Been Down’ Written by Haim & Rostam Batmanglij Produced by Danielle Haim, Rostam Batmanglij & Ariel Rechtshaid

and this lends itself to a modern sound, because these are modern tools.”

Eclectic Avenue Rechtshaid has an eclectic background. As a multi-instrumentalist, songwriter, engineer and producer, he has worked in pop and alternative rock with the likes of Adele, Diplo, Vampire Weekend, Madonna and many more (see www.soundonsound.com/ people/ariel-rechtshaid), and he’s been involved with Haim since their 2013 debut album, Days Are Gone. Batmanglij brought his own eclectic mix of influences to the table, not only having similarly diverse studio roles as Rechtshaid, but also as a result of his cultural background. Born in Washington, DC to Iranian and Persian parents, he says he “definitely grew up listening to music from all around the world. There was a lot of Persian music playing in my household, there was African music, there was folk music, there were the Rolling Stones. I think on this Haim album everybody’s influences pushed the album to be the eclectic mix that it became.” Work on Women In Music Pt. III began in the first half of 2019, without those involved actually planning to make an album, as Rechtshaid recalls. “About a year ago Danielle was kind of frustrated with the pressure of putting out records in time for touring and so on, and said she just wanted to put out one song. So we did a song called ‘Summer Girl’, and this and its video made by Paul Thomas Anderson came together really effortlessly, which created a new, looser direction for us. “We did a few more songs, still thinking of singles and not of making an album,

and then the record company asked what our thoughts were about doing an album. So the concept came in the middle, as opposed to at the beginning. The sounds and styles on this record have surprised people, and it is rewarding to get that reaction. But the album is eclectic because Haim are eclectic. It was only a matter of time before all those styles would show themselves on one record. The band are also only three records into their career, they’re still getting warmed up and getting better at what they do.”

Team Effort ‘Summer Girl’ was released on 31st July 2019, and was followed by the singles ‘Now I’m In It’ and ‘Hallelujah’ a few months later. ‘Summer Girl’ was also started by Danielle Haim in response to Rechtshaid’s cancer diagnosis (he is since in remission), and her first step was a demo she had made in GarageBand on her phone. About half the songs on Women In Music Pt. III were started with some kind of demo and then developed in the studio, and the other half was written in the studio. The latter is actively encouraged by Batmanglij. “I guess it is a common thread for records on which I have been a producer that there is a lot of writing songs from scratch in the studio. This may be because my role as a producer is always to fill the space that needs to be filled. If somebody comes into the studio and says: ‘I want to write a new song today,’ inevitably that’s something that I take pleasure in doing. “It’s part of my journey, which is that with every recording I have been a part of, I have been in the role of producer. I was the sole

producer of the first two Vampire Weekend albums, and then Ariel and I co-produced the third one, Modern Vampires Of The City [2013]. I think I was about 14 when I realised that it is what I specifically wanted to do as a career: write songs and produce them. I love making music on my own, but also working as a producer as part of a project with other people.” Women In Music Pt. III obviously fell into the latter category. Not only did Danielle Haim, Batmanglij and Rechtshaid produce the entire album together, they also co-wrote most songs with the other two members of Haim. The credits also feature an extensive cast of additional musicians, engineers and well-known mixers — among the latter are David Fridmann, Manny Marroquin, Tom Elmhirst, Neal Pogue and Shawn Everett. Batmanglij elaborates on how it was all put together. “The timeline is that we started with ‘Summer Girl,’ and then we finished the last song in January, which was ‘I’ve Been Down’. There was a vinyl release soon afterwards, on which Ariel and I mixed several of the songs, and for the digital release later [released 26th June], several more songs were sent to outside mixers. Ariel and I mixed three songs on the digital album, including ‘I’ve Been Down’. “The way a lot of the album progressed was with people splitting off into teams. Ariel and Danielle would work together, and Danielle and I, and Ariel and I, and Estee and Alana, and so on, and then we’d make final decisions about things when we were all in the same room together. The most effective work happened when we were in teams of two. The song ‘I’ve Been Down’ was a case in point.

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INSIDE TRACK A R I E L R E C H T S H A I D & R O S TA M B AT M A N G L I J • H A I M

Much of Haim’s latest album was recorded using Ariel Rechtshaid’s Scully 280 16-track tape machine.

Photo: Ariel Rechtshaid

“Danielle came into my studio one morning saying: ‘I want to write something new.’ She sat down in front of Pro Tools and sketched in a drum part using a Native Instruments Kontakt drum kit I pulled up called Waves Factory Old Tape Drums. We began writing the song very quickly with me on acoustic guitar. You can still hear that acoustic guitar part in the final recording. We wrote the verses and chorus of the song in a couple of hours, and recorded most of it. The chords for the bridge of the song where later written by Ariel and played by me on a Hammond organ, and the bridge was written on top of those chords.” Rechtshaid recalls: “One day Rostam and Danielle were together, and they put up a little drum loop, and then came up with a part of the song. We thought the album was complete, so we were like, ‘Is there still something missing from this record, is there space for another song?’ But I remember hearing the song and loving it and wanting to finish it. When Danielle came to my studio in Burbank it was just a drum loop and acoustic guitar and Danielle then played the beat on real drums, and we recorded them on my 16-track Scully tape machine.”

On Tape Women In Music Pt. III was recorded in a variety of locations, amongst them Vox in LA and the Strongroom in London, but mostly at Rechtshaid’s and Batmanglij’s studios. The two producers elaborate on their studios, and how they used them for their old/new process. Notable is that the studios of both are full of musical instruments and all manner of unusual gear, in the case of Rechtshaid tape recorders, which played an essential part in achieving the ‘old’ aspect of the sounds, as was the case when recording the drums for ‘I’ve Been Down’. Rechtshaid: “We wrote and recorded in my studio, Heavy Duty, and in Rostam’s studio, Matsor, and my tracking room, which is eight minutes from here. Sonically, we were experimenting all the time. It was like we had been preparing years for this moment subconsciously, and we used

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whatever toys were at hand. I have a record player, and I was pulling up some of my breakbeats and scratching them in. If it worked, it worked. Historically we did not have access to live tracking space quite so freely as we did on this record, so that gave us a lot more freedom. Also, I now have all the necessary gear, with microphones and a tape machine that I got working over the years, all in a bizarre setup. All these ingredients informed the record. “My tape machines are a Scully 280, a 16-track 2-inch; an Ampex 440 1-inch 8-track; and an Ampex 350 quarter-inch. I usually run a microphone into the balanced input of the tape machine, using the internal mic pre, and that sounds completely magical to me. I became obsessed with that sound. Before I got my hands on the Scully tape machine I had Scully mic pres, because they were an affordable, vibey, vintage option. I loved the way those mic pres sounded, so when I got the 280, I wanted to immediately start recording into it, but that didn’t fit with my workflow. “With the 440, for example, my signal chain is mics going into my Neve BCM10 or other external mic pres. The signals come up on a patch bay, and they are multed to Pro Tools and the tape machine. And

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then the signal goes from the repro head of the tape machine to Pro Tools as well. So I’m recording two times 8-track at the same time in Pro Tools. The moment the recording is done, I align the recording from the tape recorder with the digital recording, which means I have it in a playlist, and can continue to work in Pro Tools. “I started working with the Scully 280 when we began the Haim record, and wanted to use the mic pres on the machine. While tape recordings usually sound nicer to my ears, with the Scully things almost feel like a different performance. It sounds wildly different. I would listen to Danielle play, and then I would listen to the Scully and it instantly brought me somewhere else, like to a ‘60s drum sound. The problem is that you cannot mult after the Scully mic pres on the tape machine before the signal hits tape. So I had a bunch of mic splitters built, with good transformers, by Jensen. Instead of splitting the signal after a Neve mic pre we split immediately after the microphone. One signal goes to my Neve mic pres and then into Pro Tools, and the other goes straight to the Scully, and from there to Pro Tools.”

Vintage Vibe The Scully 280 tape recorder was used by the likes of Jimi Hendrix and the Rolling Stones. Recording onto that machine definitely added to the vintage vibe the producers of Women In Music Pt. III were after, before modernising it, of course. Batmanglij elaborates. “We used tape in a specific way, mostly for drums. Particularly the way the room is captured on the tape has a special presence. There’s noise and hiss, and especially when you use a lot of compression, it makes the drums come to life. Tape also changes the transients and makes them more pleasant. Once you get a vibey drum sound in the computer, you can do anything you want to it. But the vibe is what you have to capture initially, and that is what the tape machine

really does. ‘Summer Girl’ also started with Danielle drumming, and we recorded those drums and the saxophone on tape at Vox, where they have great tape machines.” Like Rechtshaid’s studio, Batmanglij’s facility has a combination of 21st century state‑of‑the‑art stuff and gear that can add a vintage vibe. “I have some old and new Neve mic pres, and two Tube-Tech compressors, and a Blue Stripe Anniversary Edition 1176. My soundcard is a UAD Apollo 16 and my monitors are the Dynaudio BM5As and PMC twotwo.8s, with the big sub. I bought the Dynaudios when I was 22, and I have mixed every record on them that I have ever made. They were just $500 each, but I love them and know how to make things sound good on them. The studio has superquiet air conditioning, because it’s all in one room. The noisiest thing in my computer is my Mac Pro, the black one that looks like a trashcan. ”I have several microphones, including an AKG C414, Neumann U87, two Coles mics, and an RCA KU-3A ribbon, but my main microphone is the Sony C800G. How bright it is depends on what preamps and compressors you pair it with and the room you record it in. I have all mics, mic pres and compressors always patched in, ready to record, so I literally only have to hit one key in Pro Tools to record whatever instrument. I’m also a big fan of using DI, and for the Haim record I got the Tonecraft 363 tube DI, which is amazing.”

The DI’d bass enjoyed a simple signal chain comprising just a FabFilter compressor and Pro Tools’ Air Distortion plug-in.

edge. This process of in-the-box treatments was in effect rough mixing, and they took this to a point where they didn’t really need to do final mixes. Rechtshaid: “Many people source loops and samples, and they make songs around that. We were making songs and then designing sounds that have the right vibe for that song. This is part of production, and also of mixing. So much of mixing is in the production these days, and at the end of that process there often was no purpose to doing another mix. When the song is right, and the production is right, it becomes like a choice rather than a necessity to send it to an external mixer. So in effect Rostam and

I were mixing as we went, passing songs back and forth to each other.” Batmanglij: “There were times when Ariel and I would sit together in front of a session, but we did a lot of the work separately from each other, and then we’d trade sessions. We synchronise our drives every two or three days using SyncPro, and we easily can open up each other’s sessions in our computers. The reason that process works is because we like the things the other does. Ariel and I also both try to get things to sound as close to a finished recording from the very beginning. With anything we sent to external mixers we would have done our best to make it sound

Chop & Change Batmanglij and Rechtshaid did a lot of the engineering themselves, and spent many hours chopping and editing vintage-sounding recordings and then adding all sorts of plug‑ins to add a contemporary Drum bus processing on ‘I’ve Been Down’ included FabFilter’s Pro-Q 3, PSP’s Vintage Warmer, Waves’ Chandler TG12345 emulation and Oeksound’s Soothe.

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INSIDE TRACK A R I E L R E C H T S H A I D & R O S TA M B AT M A N G L I J • H A I M

Arturia’s Wurli plug-in provided the electric piano sound.

as good as possible. We wouldn’t send out unfinished mixes. But in the case of ‘I’ve Been Down’ we were entirely happy with the way our mix sounded.”

Drums & Bass Batmanglij: “There’s quite a bit of editing on the drum tracks, because we love drums that have a combination of natural and unnatural. So we preserved Danielle’s feel, but we also cut things up to add a quality that’s hip hop influenced. It makes it sound like a loop of an old soul record that has been chopped up. We were also arranging the drum part. While recording drums we asked Danielle to do one take of fills, and we later created a drum performance using these fills.” Rechtshaid adds: “The drums were recorded very simple. Most of the drum sound comes from

the one RCA 44. There’s also a Neumann FET 47 on the kick, and an STC 4033. They gave us a bit more low end. But 95 percent of the sound came from the RCA 44, which has a very Rostam move on it, in that he dipped 6.45kHz. He is very sensitive to what he calls harsh frequencies.” Batmanglij: “When working on ‘Summer Girl’ I used the Waves Sound Shifter plug‑in to pitch the drums down 4-5 semitones, and that added quite a bit of grit. I did it again with the drums on this song, pitch‑shifting

The saxophone was heavily processed, with multiple EQs, delay, formant shifting and a convolution reverb using an impulse response captured by Batmanglij at his own studio.

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the drums up two semitones. I was just experimenting to see how the ring of the snare sat in the mix. Also, pitch‑shifting makes things sound more sampled, of course. I sent this to Ariel, and the next time I got the session back, he’d replaced the Sound Shifter with the Serato Sample plug-in, doing the same pitch‑shifting, but with a different sound. “The drum bus has the FabFilter Pro-Q 3 dipping around 200Hz, the Waves TG12345 for compression and EQ, the PSP Vintage Warmer and the Oeksound Soothe. The Rolling Stones were a touchstone for us in this song, and the Vintage Warmer is on a semi-driven tape setting adding saturation. Because the drums were compressed heavily, and also because of the pitch‑shifting, we got some unlistenable harsh frequencies, which were taken out by the Soothe.” Next, Batmanglij recalls the bass recording: “There’s a Fender Jazz in the bridge, and the rest of the song has a Hofner bass. They were both recorded using the Tonecraft DI.” “We used the FabFilter Pro-C 2 compressor and the Pro Tools Air Distortion on it. It is very simple,” adds Rechtshaid. “We did not do much fancy stuff like parallel compression and stuff like that. It is great for certain things, but not for this song. This song was emoting immediately, and it was about not losing that.”

Wurli Batmanglij: “We wanted to have the sound of an old Wurlitzer, recorded to tape in a dusty studio. It enters in the bridge. I rarely do this, but I created a Pro Tools

track preset in a different session, and then used it here on the Wurli track. The Wurli V2 plug‑in is made by Arturia, and they achieve the sound with synthesis, as opposed to sampling, and it sounds very real to me. None of the pedals are used, but the reverb is activated. “Then there’s the Aberrant DSP Sketch Cassette, a plug-in made by two young guys in upstate New York. It just cost $20, and I think it is one of the best plug-ins that has ever been made! It adds a very vintage vibe, and has something called NR Comp, which is a very aggressive multiband compressor that really changes the sound. After that there are the Goodhertz Vulf Compressor, and the Goodhertz Wow Control for yet more wow and flutter. All this was about retracing the steps of an old or sampled recording. In the old days people recorded to tape and then they had to bounce tracks down to other tapes, and when you do that you are having tons of different wow and flutter enter the signal. My goal here was to recreate that situation.”

Saxophone The saxophone was heavily processed, as Batmanglij explains: “Yes, there are a lot of plug‑ins on the sax! It was recorded with the Sony C800, and played by Henry Solomon, who has a new sound that was integral to this album. There are a Pro-Q 3 and a Waves RCompressor and the Avid EQ3 7‑band doing a low-cut, and the Soothe to take out harshness. But the main plug‑ins are the Soundtoys Little AlterBoy, the Altiverb, and the UAD Helios. I love formant shifting the saxophone. The formant shifting from the Little AlterBoy makes it sound more sampled, again, and also much darker. I am also using the distortion that is built in the Alter Boy. “The UAD Helios EQ changes the sound quite dramatically. The most fascinating plug‑in on the sax comes after that and is the Altiverb, because it uses a sample that I made myself at Vox, by putting my Dynaudio BM5s in the live room there, and recording the impulse response. That is part of how that saxophone sounds so alive. I am already recording the sax from about three feet away, and then adding more room with the Altiverb. Finally, there’s the Slate Digital Repeater set to an 8th‑note delay, just a touch to bring it into the realm of the Rolling Stones in the ‘80s.”

Acoustic Guitar Batmanglij: “I recorded my acoustic guitar with the two microphones that I normally

a nonlinear reverb. The nonlin is the main feature on the vocal sound. I am a big fan of using the nonlinear on vocals just to give a little bit of a 3D quality. The version that I like the best is the Altiverb AMS RMX16 nonlin. I have the real unit, but I also like the way that Altiverb version sounds. “There’s a different vocal sound in the bridge. The plug‑ins are again the Waves Q2, taking out everything below 60Hz and dipping around 250Hz; the CLA76; and then there’s an echo panned all the way to the right, coming from the UAD EP-34 Tape Echo. The dry vocal is in the centre. More space is added with a send to a chamber reverb aux, which uses the Cello Studio in Altiverb. We also had a send to the Ocean The acoustic guitar was treated with some EQ and a dose of Sketch Casette. Way aux for the have on my piano, an upright Steinway chant in the bridge. That aux has the UAD K52. The mics were about two and a half Ocean Way Studios plug‑in, as well as the feet apart from each other, and both facing Cranesong Phoenix II, and the Massey the acoustic guitar. By not having the mics L2007 limiter for more loudness. The idea so close to the guitar you get a lot of room is to make the chant sound like a handful sound. Trying to get more room sound has of people singing in a room, even though always been my approach to recording they were recorded individually.” acoustic instruments. I added the Waves Master Bus Q10, using a preset that I made to get rid of the low mids.” Batmanglij: “The UAD SSL G-Series Bus Rechtshaid adds: “When Rostam and Compressor is grabbing a few peaks. The I were doing Modern Vampires In The iZotope Ozone 9 is not doing very much, mostly just dynamic EQ, taming some City, we were obsessed with the UAD frequencies that had built up, and allowing ATR102 plug‑in, because it is adding EQ, us to have a loud recording that isn’t compression, modulation and distortion. harsh. The Waves L2 Maximizer is the final During the making of this album, Rostam plug‑in, and you either like it or you don’t got the Sketch Cassette plug‑in, and we like it. I tend to like it! first used it on the acoustic guitar. We were “As we mentioned, the touchstone going for Keith Richards’ acoustic guitar with this song was the Rolling Stones, sound in the Stones’ ‘Street Fighting Man’. and in the bridge I was thinking of Phil That plug-in gave us the essence that we Spector, but updated. That is always were looking for. It’s great.” floating around in the back of my Vocals mind during the making of the entire album: how do we make something “Danielle’s vocals were recorded with that touches on those older recording the Sony 800, and a Neve 1073LB and techniques and effects, and also make it my 1176AE, black with the blue stripe,” sound modern and competitive? Strict Batmanglij explains. “There’s not much recreation is less inspiring to me; I want on the lead vocals, other than the Waves the recordings that I am a part of to Q10 EQ, the UAD 1176AE in the verses sound of a new era.” and Waves CLA-76, and then a send to

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ON TEST

Apple Logic 10.5 Digital Audio Workstation

Logic 10.5 introduces a raft of new features and remains exceptional value for money. MARK WHERRY

I

always get a slight frisson when Apple release a major new version of Logic Pro X. Part of this can be attributed to nostalgia, but mostly it’s because Apple never pre-announce such a release: it just shows up on the App Store, accompanied by a press release and an updated web page. This time, however, a major new release wasn’t quite so unexpected, thanks to a screenshot that had briefly appeared on Apple’s web site at the end of March. The screenshot in question showed a version of Logic featuring what looked like two GarageBand-inspired features — Live Loops and Remix FX — portending what could be expected in a forthcoming upgrade. Despite the toothpaste being out of the tube, Apple remained silent until May 12th when I found myself downloading

Apple Logic 10.5 £199 pros • Live Loops seamlessly integrates modern, loop-based music creation into the existing Logic workflow. • Sampler and Quick Sampler reinvigorate Logic’s sampling abilities. • Step Sequencer offers abundant functionality and is just plain fun. • Live Loops and Remix FX can be performed via multitouch gestures using the Logic Remote app.

cons • It’s churlish to complain about much at this price. But... • The Piano Roll editor remains less sophisticated than some of the competition. • Serious surround work doesn’t seem to be a priority.

summary Logic Pro 10.5 is a comprehensive upgrade, with new functionality, workflows, samplers and content. And, as Apple suggest, it embodies some of the most significant changes to the application since the introduction of Logic Pro X.

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the newly released Logic Pro 10.5 from the App Store. As with previous upgrades since the launch of Logic Pro X in 2013, it was once again free for existing users, and newcomers will be charged the very reasonable price of £199.99 for the entire package. You just need a relatively good Mac running at least Mac OS Mojave 10.14.6, with 6GB of free storage for a minimal installation; with Logic Pro 10.5 boasting 2500 new loops, 70 extra kits, and over 1500 additional patches, the full set of content requires an installation with 72GB of storage space. It’s worth noting that when Apple recently extended the free trial period of Final Cut Pro X (the company’s video editing application) from 30 to 90 days, they also introduced a free trial of Logic Pro X for the first time. This can be downloaded from the company’s web site (as opposed to the App Store) and currently expires after the same generous period. Apple recommend that existing users should back up both their project files as well as the currently installed version of Logic Pro X before updating, and this is exceptionally good advice. Whilst backing up work is always prudent, it won’t be possible to download previous releases of the application from the Mac App Store once you’ve updated to 10.5. You can archive the existing version by right-clicking the application’s icon and selecting Compress from the contextual menu, storing the resulting .ZIP file (which you might want to rename) in a safe place in case you need to revert to this earlier version, or move it to another Mac.

It’s Alive As anticipated, the headline new feature in Logic Pro 10.5 is indeed Live Loops, first seen in GarageBand 2.1 on iOS back in January 2016. However, as you might expect, Logic Pro X proffers a more sophisticated implementation, aiming to combine a second, nonlinear approach to music creation within a single, Logic-based workflow. Traditionally,

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Logic has used a linear method of timeline-based sequencing, while the new, nonlinear approach invites an unavoidable comparison to Ableton’s Live, enabling you to work from a collection of ideas in a more free-form manner without worrying about a timeline. To make this possible, the Project window can now display two different views, either together or individually. There’s the traditional Tracks Area we all know and love, where regions of musical data are represented by rectangles based on their musical length, and the new Live Loops grid, where individual musical ideas that generally work well as loops are stored as cells within that grid. A vertical line of

cells in the grid is referred to as a scene, and each horizontal row plays a cell back via the corresponding track in the track list (or ‘Track Header area’ in Logic parlance). This is particularly neat, since it makes it straightforward to share ideas between the two areas, as we shall see. Getting started with Live Loops is easy, thanks to some new options in the Project Chooser window. By selecting New Project in the list on the left, you can either create an Empty Project as before, or you can start a new Live Loops Project, where the Live Loops Grid is shown instead. Conceptually, cells are to the Live Loops grid what regions are to the Tracks Area, and in the same way you can have

different types of regions — such as audio, MIDI and Drummer — the Live Loops grid also has the equivalent cell types. Empty cells are displayed as blank, dark grey rectangles, while a cell containing musical data is represented by a colourful square. A cell looks a bit like a region, in that its name is shown at the top above a visual overview of the content, this time in the form of a circular graphic representing the Loop Length of a cell. A Play/Stop button appears when you hover the pointer over a cell, which can be clicked to start a cell playing. During playback, an animated pie slice represents the playback location, and you can stop a cell by once again clicking its Play/Stop

button. Alternatively, you can also stop a cell by clicking the Play/Stop button in an empty cell on the same row, or by clicking that row’s Stop button located on the far right of the Live Loops grid. Note that although you can play back cells on different rows simultaneously, only one cell per row can be playing at any given time. Starting a cell on the same row as another cell that’s playing will switch playback to the newly started cell, ceasing playback of the previous cell. To start all cells within a scene, click the trigger button for that scene depicted by a ‘hat’ symbol at the bottom of the Live Loops Grid. To stop the playback of a scene (or any combination of cells),

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you can click the Grid Stop button at the bottom right of the grid. As you start and stop cells playing, you’ll probably notice that a cell doesn’t necessarily start or stop playback immediately when you click its Play/Stop button. This is because Logic waits for the cell’s next Quantise Start time before carrying out the action, enabling cells to be started and stopped in time with other cells that might be playing. If a cell has a Quantise Start time of ‘1 Bar’, for example, it will always start or stop playing on the next bar position. Each cell can have its own Quantise Start value, although a cell will use the grid’s Quantise Start time if its own parameter is set to Global. To set the grid’s Quantise Start time, simple click the setting in the right-hand side of the Live Loop grid’s toolbar. And you can also assign a Quantise Start time to a scene, which will be used by all the cells in that scene when it’s triggered, overriding a cell’s own setting. To do this, right-click a scene’s trigger button to open a contextual menu and select the time required from Quantise Start submenu.

Queue On Cue Starting and stopping cells during playback as just described is known as queuing, although ‘cueing’ might seem more appropriate. If you want to queue a cell for playback, simple right-click a cell and choose Cue Cell Playback from the context menu. You’ll notice how a cell that’s queued for playback blinks to indicate it will start playing when you engage Logic’s normal playback, such as by pressing Space. If you stop the playback by using Logic’s transport (pressing Space again, for example), playback stops immediately, as opposed to waiting for the next Quantise Start position, but the queued cell remains queued. You can dequeue a cell by right-clicking it and selecting Dequeue Cell from the context menu, or clicking the Grid Stop button. Queuing a single cell for playback probably isn’t that useful, so it’s possible to select multiple cells simultaneously and add all of them to the queue. To do this, select a cell by clicking its name at the top part of a cell, and then, holding down Shift, select other cells in the same manner. Using the Cue Cell Playback command from the context menu will queue the selected cells, or you can use the handy Option-Return key command. Selected cells can be removed from the

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EXS24 metamorphoses into Sampler in Logic Pro 10.5, sporting a modern user interface that incorporates playback and editing parameters in a single, resizeable plug-in window.

queue by choosing Dequeue Selected Cells from the context menu or by pressing Option-Return again, as well as using the Grid Stop button, or by using Logic’s Stop command multiple times by pressing Enter. As you might expect, you can queue and dequeue a scene rather than individual cells by right-clicking a scene’s trigger button and choosing Queue and Dequeue Scene from the context menu. And, again, you can dequeue a scene using the Grid Stop or standard Stop commands. Following typical Logic conventions, cell parameters for a selected cell can be adjusted in the Cell Inspector, and you’ll find region-like settings you’ll already be familiar with for muting, transposing and quantising. These take their place alongside settings for configuring playback, the loop, length and speed of

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a cell, as well as neat reverse option that plays a cell backwards. Cells can be created manually by right-clicking an empty cell and selecting the appropriate option from the contextual menu. If you right-click an empty cell on an audio track/row, you can select Add Audio File and choose the file from the Open dialogue; alternatively, you can drag in an audio file, as such one from the Library. Should the audio file in question already contain metadata with loop information, this will be used by the new audio clip. Similarly, if you want to program your own cell using note data, right-click an empty cell on a software instrument track and select Create MIDI cell. Like MIDI regions, MIDI cells can be edited in one of the editors, and double-clicking one opens the Piano Roll editor by default.

If you want to create a cell that can be programmed using Logic Pro 10.5’s new Step Sequencer (which we’ll cover later in this review), you can create a pattern cell instead. And finally, as you might expect, if you have a Drummer track, right-clicking an empty cell in the corresponding row provides access to a Create Drummer Cell command from the context menu. It’s also possible to record your own audio and MIDI cells, starting with an empty cell or by creating a cell and recording into it. The advantage of the latter approach is that you can configure a cell’s parameters before recording, and, indeed, a cell will be created for you in any case if an empty cell is selected and you start adjusting its parameters. You can start recording by clicking a cell’s Record button, which appears instead of a Play/Stop button when a track is armed, or simply press Option-R on a selected cell. The length and behaviour of the recording for a cell is set in the Cell Inspector, and one particularly neat option is to set the Cell Recording mode to Takes, so that a new cell is created every time there’s a loop, which becomes a take cell once you press Stop. You can switch between takes as normal, or — and this is the cunning part — unpack the takes to different cells by right-clicking the take cell number and choosing a command from the ‘Unpack Take Cell to’ submenu. For example, selecting Next Empty Cells Right (or pressing Ctrl+Command+U) creates new cells in scenes to the right of the take cell, whereas choosing New Scenes creates new, empty scenes that the additional takes are placed into.

You Spin Me Right Round, Logic As I’ve already hinted at, one of the best things about Live Loops is the way it can be used — quite literally — alongside the traditional Tracks Area. Both can be visible simultaneously when the Show/Hide Live Loops Grid and Track Area buttons are enabled, and a divider line separating the two areas can be dragged to see more of one and less of another. And because the track list is persistent between the Live Loops grid and Tracks Area, with each row in the grid corresponding to the equivalent track, you’re working with the same set of tracks in either area. This split area view makes it possible to drag one or more regions from the Tracks Area to cells in the Live Loops Grid, and this works the other way around as well,

Remote Loops Given that a mouse or trackpad isn’t always the most tactile way to trigger Live Loops, Logic supports Novation’s Launchpad controllers, automatically mapping Live Loops to the pads and other controls. Since I don’t own a Launchpad, I wasn’t able to try this for myself. However, it’s also possible to trigger Live Loops from the newly updated Logic Remote app for iOS, which is freely available on the App Store and works with either an iPhone or iPad running iOS (or iPadOS) 13.1 or later. This worked brilliantly, especially on an iPad where the larger display lets you simultaneously control the master Remix FX, enabling a fluid, multitouch performance.

The Logic Remote app, seen here running on an iPad Pro, makes it possible to control features such as Live Loops and Remix FX from a compatible iOS device.

so you can drag a cell from the grid to the Tracks Area to create a new region. Copying between the two areas in this way creates independent copies of cells and regions, so tweaking a region you’ve copied to a cell doesn’t affect the cell that was created, and vice versa. Multiple cells can be copied at once, or you can copy or insert a scene to the playhead position by right-clicking the appropriate scene trigger and selecting Copy or Insert Scene to/at Playhead. Creating regions from cells is relatively straightforward, whilst you can take full advantage of the Track Area’s editing tools when producing a cell from a region. For example, in addition to dragging a region into the grid, you can use the Copy Region Selection to Live Loops command from the Track Area’s Edit menu to make a new cell on the appropriate row from a selection, which can be a simple region selection, or, for example, copied from a marquee selection.

The logical culmination of this workflow is the ability to record a performance in the Live Loops grid into a standard arrangement within the Tracks Area. To do this, ensure that Cycle mode is disabled, click the Enable Performance Recording button (on the left side of the grid’s toolbar) so that it turns red, and start recording in the usual manner by clicking the Record button. Now, dressing in your DJ attire with the optional smoke machine and strobe lights activated, let the performance commence, stopping Logic’s transport when you’re done. There’s a good deal more to say about Live Loops, especially since it’s one of those features that’s simple to get started with, but offers a broader complexity the deeper you delve. For example, it is possible to use Smart Tempo and transients on audio cells, as well as other advanced functions. And although Apple suggests Live Loops is ideal for those making music in such genres as EDM and

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The new Quick Sampler makes it possible to create a new sample-based instrument from a single audio file. Here we can see a drum loop with markers that have been created from transients, enabling each slice to be played independently.

hip-hop — which may be true — I actually think this nonlinear way of working is going to be useful for those working in pretty much any genre.

Say Hello, Wave Goodbye Something that gave me a tinge of wistfulness in Logic Pro 10.5 is the sun setting on the EXS24 sampler; after 20 years of service, you’ll no longer find it in the list of available instruments. I remember the excitement of receiving Emagic’s Xtreme Sampler 24-bit in the summer of 2000 and being blown away by its simplicity, integration with Logic Audio 4.3, and efficiency compared with other software samplers at the time. However, change is inexorable, and replacing EXS24 is a new instrument called Sampler — how do they come up with these names? The first thing to affirm about Sampler is its backwards compatibility with EXS24. It uses the .EXS file format natively for both reading and writing instruments, and when you load a project created with an earlier a version of Logic using EXS24 plug-ins, these instances will be automatically reassigned to Sampler with the same instrument and settings loaded as before. At the top of Sampler’s interface is the Navigation bar, where you’ll see a group of buttons. The first three — Synth, Mod Matrix and Modulators — give access to familiar playback parameters, while the last two — Mapping and Zone — provide instrument-editing capabilities for adjusting Zone settings and specifying how those Zones are mapped. Clicking a Navigation button scrolls to the appropriate pane, and a small yellow ‘LED’ indicates whether a given pane is visible in the interface — clicking it toggles the visibility of the corresponding pane. You can also Option-click a Navigation button to show only that pane (hiding all others) or double-click it to expand a pane to fit the available vertical space. In reorganising the interface, there are a few things worth noting for existing Logic

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Pro users. Firstly, certain global settings previously accessed via EXS24’s Options and Virtual Memory windows are now located in the Audio pane’s Sampler tab in Logic’s Preferences window, which makes sense. The commands for importing DLS, SoundFont and Giga-format instruments have been removed, although, for the longest time, all these options have done is throw up a message box telling you where to place the file to be imported. You can still place files at this location (~/Music/ Audio Music Apps/Sampler Instruments) to be imported as before. And finally, a familiar Actions pop-up is available with additional commands concerned with initialising and mapping the synth parameters found in the Synth, Mod Matrix and Modulators panes. While the playback features remain largely as EXS24 users have come to expect, albeit now made more accessible by the cleaner façade, there are a couple new engine features that merit a mention. An instrument can now use up to five envelopes and four LFOs, each offering a full set of controls, with the LFOs offering monophonic or polyphonic triggering. You can add or remove envelopes and LFOs via the +/- buttons at the top of the Modulators pane, although one envelope must be present in each instrument. In addition to extra modulators, Sampler also includes a second filter stage

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that can be enabled or disabled like the first. The filters run in series by default, although it’s also possible to use them in a parallel configuration where the balance of the output signals can be set with the Filter Blend knob, which can be modulated to crossfade between the filters.

The Creation Myth Playback advancements aside, arguably the biggest improvements in Sampler come from a more streamlined approach to instrument creation and editing. As an example, there’s no longer a separate window for editing instruments; the Mapping and Zone controls are now included as panes within the main interface, allowing for a simpler workflow. Starting with an empty Sampler instance, you can drag audio files and regions onto the Navigation bar to build your instrument. When you drag something over this bar, it displays two dropzone areas: Chromatic and ‘Optimized’. Chromatic will map the samples to keys chromatically from C2, using the original looping, length, tuning and loudness settings from the audio file. Optimized, on the other hand, will map samples to keys based on pitch analysis, crop any silence, search for loop points, and adjust the tuning and loudness as required, courtesy of some Redmatica-inspired goodness. These tasks

can also be carried out after a zone has been imported if you’d rather start with the default, vanilla options. When one or more items are dragged over the Chromatic area, it splits into two further dropzones specifying either Zone per File or Split at Silence. If the audio is dropped on to Zone per file, a new group will be created containing one zone for each file; if the destination is Split at Silence, a new group is created for each file, with zones being created within each group representing audio segments split between silence from the appropriate file. Similarly, if the Optimized dropzone is used, this also subdivides into two dropzones: Zone per File, as just described, or Zone per Note, where a group is again created for each file, but this time with zones being created and mapped based on the pitch analysis of each note in a file. This represents just one method to create groups and zones. You can also create them manually through the Group and Zone menus in the Mapping pane, assigning audio files to zones as required,

Re-re-remix FX Logic 10.5 includes a new Multi Effect called Remix FX that, as with Live Loops, is also based on a similar feature in GarageBand. In fact, it looks and sounds pretty much identical, with the exception that it can be used like any other effect in Logic on any track and not just the master output. Remix FX features two Kaoss Pad-like vector controls, with each addressing one of six available effects: Filter, Repeater (for stutter-like effects), Wobble (an analogue-esque filter with modulation), Reverb, Orbit (a flanger and phaser) and Delay. The horizontal and vertical axes control parameters based on the selected

effect, so with Filter chosen, for example, the X and Y axes modulate the cutoff frequency and resonance amount. Each effect also has an extra parameter accessed via the settings button; in the case of Filter, this selects between ‘Phat’ (24dB) and ‘Classic’ (12dB/octave) modes. The centre section of Remix FX has two faders for gating and down-sampling effects (the latter similar to Bitcrusher), along with buttons for manipulating the playback of incoming audio. You can reverse the playback, add a vinyl-inspired record scratch, or emulate a tape stop effect with the requisite sloowiiiiing down effect.

or by dragging audio into the Zone pane or onto a key or range of keys in the Mapping pane’s keyboard view. Adding and removing audio files to and from zones can be achieved just as easily with a similar variety of methods. Since the concept of Articulations was added in Logic Pro 10.4, it makes sense for Sampler instruments to support this functionality in a similar way to other instruments like the Studio Horns and

Strings plug-ins, which were also included in the previous update. Articulations are specified at the group level with an ‘Enable by Articulation’ parameter: simply switch it on and specify an Articulation ID value (starting with 1) to which that group should belong. When an instrument is loaded that supports Articulations, you can select New from the Articulation Set pop-up in the Track Inspector, which will create and use an Articulation Set based on the

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Sampler instrument. It would, of course, be convenient if there was a way for this step to be carried out for you. Despite the nostalgic tone used earlier to describe EXS24’s retirement, it was admittedly a plug-in that harkened back to an earlier era and, if one is being honest, an improved replacement was probably overdue. L’EXS24 est mort, vive l’EXS!

Quick Quick Slow For those whose sampling needs are, dare I say, simpler, a second new sampler instrument called Quick Sampler is also included. This is conceptually redolent of Live’s Simpler and Cubase’s Sampler Track. Reminiscent of the early days of sampling, Quick Sampler creates a sampled instrument from a single audio file; and, unlike most so-called samplers, actually allows you to record a sample directly into the plug-in from a hardware input or the internal output of a software instrument or bus. Alternatively, you can drag an existing file or region into the waveform display rectangle in the upper section of the plug-in’s interface. And it’s even possible to use a MIDI clip or region if its output is a software instrument, prompting Logic to automatically render an audio file to be used as the basis for a new instrument. When you drag something to be imported into Quick Sampler the upper section shows two dropzones, similar to Sampler, so you can decide whether to import the original audio as is, or whether it should be ‘Optimized’. Since the choice is non-destructive, you can later revert to either the original or optimised sample by using the Reimport Original or Reimport Optimized commands from the plug-in’s Action menu. Any audio file that’s used by or created with Quick Sampler is stored within a Project. If you’re saving a Project using folders, for example, a Samples folder is created within the Project hierarchy that saves independent copies of existing samples and any recordings made with the plug-in’s Recorder mode. Quick Sampler offers three modes of playback: Classic, One Shot and Slice. Classic mode plays back the sample for as long as you hold a key and enables loop points to be specified, whereas One Shot plays the entire sample when triggered and obviously doesn’t permit looping. In either mode you can set whether the sample should be played forwards or in reverse, which is a nice touch.

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Drum Machine Designer has been updated in Logic Pro 10.5. In the kit shown, the Kick 1 pad triggers a Quick Sampler instance, the editor for which is displayed inside the Drum Machine Designer window. The upper section is shown by default when Q-Sampler Main is selected, and the lower section is accessible when you click the Q-Sampler Detail button.

Slice mode places markers within the sample to indicate slices of audio, such as different beats within a drum loop. Each slice is mapped to an ascending keyboard range, beginning from the specified Start Key, and is played back for the full length of the slice when triggered. You can invoke Classic-style playback by enabling the Gate parameter, or instruct Slice mode to continue playback past the triggered slice to the end of the sample by toggling the ‘Play to End’ button. The lower section of Quick Sampler’s interface provides access to synth-style controls used for playback, consisting of two LFOs and a simple four-way modulation matrix, along with pitch, filter and amp stages, each modulated with a dedicated envelope. Different envelope configurations are available, although envelopes containing a release stage are only available when using Classic mode. And it’s particularly fun to modulate the sample or loop boundaries for some interesting variations. Normally, in Classic or One Shot modes (with Key Tracking enabled in the pitch pane, as it is by default), the pitch of the sample is adjusted based on the root key by speeding up or slowing down its playback rate. However, Quick Sampler also features a Flex mode, enabling a sample to sound with the appropriate pitch adjustment without changing the playback speed. You can also set

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a multiplier for the playback rate, such as half or double speed, for example, and specify whether the sample playback follows the Project’s tempo, which is useful for gapless playback in Slice mode.

Designer Drums Drum Machine Designer, which was introduced in Logic Pro 10.1 for managing electronic drum sounds, has been revamped in Logic Pro 10.5. It now uses Quick Sampler for audio samples and Drum Synth (see box) as the default instrument for synthesized drum sounds instead of Ultrabeat. As before, adding Drum Machine Designer to a Software Instrument track turns that track into a Track Stack where each drum has its own sub-track within the master track, and you can change the plug-in used by each of these tracks to any instrument plug-in. The interface looks largely the same at first glance, offering 48 drum pads split across three pages. You can mute and solo a pad’s sub-track using the M and S buttons as before, but the MIDI input and outputs for each pad are now user assignable. For example, if a pad’s input is set to C1 and its output is C2, notes played with a pitch of C1 on the master track will trigger a sound with a C2 on the appropriate sub-track. This means you can now have multiple pads being triggered from the same input note, amongst other possibilities.

Slip Rotate It Two of my favourite new editing capabilities in Logic Pro 10.5 allow you to slip and rotate content in regions, courtesy of new commands hidden in the ‘Edit / Move’ sub-menu, which are more usefully assigned as key commands. Whereas the existing Nudge Left and Right commands move an entire region left or right by the nudge value, Slip Left and Right (Ctrl+Option-Left/Right) moves the contents of a region left or right, whilst leaving the region boundaries where they are. This is useful where you have content that exists outside of the current region boundaries, such as when an audio region plays back just a bar or two of a larger audio file, and you want to ‘slip’ that content within the region. The Rotate commands are perhaps less obvious. What does it mean to ‘rotate’ audio and MIDI data within a region? With the nudge value set to an eighth note, for example, selecting Rotate Left (Ctrl+Option+Command-Left) will shift the content left by an eighth note and ‘rotate’ what was in that first eighth note around to be the last eighth note in the region. For a MIDI region this moves notes in the existing regions, whereas rotating an audio region creates a folder containing two regions that consist of the front and back parts of the original region. This feature is a great idea, although I noticed some odd behaviour when rotating MIDI events. For example, if I have three-bar MIDI region that I resize to be two bars in length, rotating left affected MIDI notes beyond the region end point rather than only the notes within the region boundaries. I’m not sure if this is a bug or a feature, but the result seemed counter-intuitive compared to my understanding of what should happen.

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Thru-5 and Thru-25 also available Hovering the pointer over the kit pad reveals new mute and solo buttons for the plug-in’s master track, along with an Actions button that reveals more commands than the previous right-click menu. Some of the new options include being able to toggle the selection of a pad by key input, or a sub-track from a pad, as well as re-ordering pads either chromatically or according to the GM drum standard. When you open Drum Kit Designer, you’ll be presented with an empty kit. As before, you can load an existing kit or assign individual sounds to pads from the Library, or by dragging and dropping audio. One particularly neat feature is the ability to drag a number of audio files or regions to an empty spot in the track list, whereupon Logic will ask if you want to create a new track using either Sampler or Drum Machine Designer. If you choose the latter, a new Track Stack will be created with an instance of Drum Machine Designer, and each audio element will be assigned to a pad within a new kit.

Step In Time Until now, there have been three basic region types you can use as containers for different kinds of musical data in Logic Pro X: MIDI, Audio and Drummer. In Logic Pro 10.5, there’s a fourth type of region referred to as a Pattern region, which contains notes and automation data that can be edited in the new Step Sequencer. Pattern regions are coloured yellow by default, showing an overview of active steps when viewed with sufficient height. And if you’re working in the Live Loops grid, it’s also possible to use and create the equivalent pattern cells that function in much the same way as pattern regions.

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Logic Pro 10.5’s Step Sequencer is accessible in the Editors pane or in a dedicated window (shown here) and can be used to program cells or regions containing patterns of both notes and automation.

To create a pattern cell or region, you can right-click an empty cell in the Live Loops grid or an empty spot in the Tracks Area and select Create Pattern Cell/Region, which can be assigned to a key command. Double-clicking a pattern region will open it in the Step Sequencer tab of the Editors area. The Step Sequencer pattern contains a series of rows, each comprising the number of steps defined by the Pattern Length menu on the Step Sequencer’s toolbar. The lengths range from 12 to 64 steps in varying increments, and although the ability to use an arbitrary number of steps would have been nice, there is a workaround by setting the loop start and end parameters for each row. The step length is specified by the Pattern Step Rate menu, located in the toolbar above the row headers list, offering the usual Logic variety of straight, triplet and dotted values.

Step Sequencer works seamlessly with Drum Machine Designer, and using this plug-in is a good way to become familiar with Step Sequencer, since pattern regions on Drum Machine Designer master tracks automatically give you a set of rows based on the notes of the chosen kit. To toggle steps on and off, make sure the Edit Mode selector on the Step Sequencer’s menu bar is set to Step On/Off on the left side, and then click a step to turn it on or off. As with the Step Editor, you can click to turn a step

Little Drummer Synth Where Quick Sampler is ideal for playing back sampled drum sounds, another new plug-in called Drum Synth is perfect for creating those retro-esque, electronic equivalents. While many presets are included, creating and tweaking your own hits is easy. Start by using pop-up menus to select the group Drum Synth is a new instrument for generating electronic drum type (either Kicks, Snares sounds. The simple interface lets you pick a type of drum, and then and Claps, Percussion or adjust the tone with a number of straightforward controls. Hats and Cymbals) and sound type (which depends on the chosen sound, whereas the Kicks group has a Shape group) to be generated by the synthesis engine. parameter to adjust the shape of the kick The icon will change accordingly (which can be drum generated, from a small, tight sound, to clicked on to trigger the sound), and a number a larger one with a boomy resonance. of parameters will be shown, such as pitch, Drum Synth a great little plug-in that’s tone, key tracking, and whether to use Mono, luculent and simple to use — especially Poly(phonic) or Gate modes for playback. compared to Ultrabeat — and sounds crisp Each group also has number of specific and clean, and ready for further devilish parameters. For example, the ‘Snares and processing. It seems fittingly at home when Claps’ and ‘Hats and Cymbals’ groups have used within Drum Machine Designer, as a Body parameter to adjust the depth of the though it should always have been there.

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on and then drag the mouse wherever you like to activate other steps. The same technique works when clicking to disable a step and dragging to deactivate other steps. To adjust the values of properties associated with a step, enable the right side of the Edit Mode selector, choosing the property to be edited on each row header from the pop-up menu. The default is ‘Velocity/Value’, which, unsurprisingly, lets you adjust the velocity of notes by clicking the desired step and dragging vertically — you can also drag horizontally to adjust the velocities of multiple steps in one pass. Properties like velocity can be adjusted for a step whether that step is on or off, although active steps appear in a brighter hue. A nice touch is that you can Command-Click when adjusting most properties to turn steps on or off without having to switch edit modes. In addition to velocity, the right-side edit mode also allows you to set values for gate (to shorten the note length triggered by a step), tie (adjusting a step length by tying it to the next and/or previous step), skip (setting that step to be skipped during playback), note repeat (dividing a step so that it triggers multiple times), chance (sets the probability of a step being triggered), loop start and end, and more. As well as using the Edit Select Mode to set what’s being edited on the row headers, you can also use multiple subrows to view and edit parameters for each row, a bit like using automation tracks in the Tracks Area. To make this possible, each

row has a disclosure triangle that can be toggled to reveal subrows. Two subrows are available by default, and you can add or remove subrows as desired. In addition to creating steps that can play back notes, you can also assign rows to trigger automation parameters as well, either in the same pattern or in a separate pattern region. This means that because pattern regions can be created on audio tracks, you can automate an audio channel by creating a pattern region on an additional audio track assigned to the same audio channel as the track playing back audio regions. This only works with regions (which is to say: not cells), but is a pretty neat way of modulating any parameter that can be automated (including those on insert plug-ins) to create some interesting textures, especially if you create a random set of data for a given row as a start point. A pair of particularly neat commands are Randomize Row Values and Randomize All Values, which can be selected from the Functions menu on the Step Sequencer’s menu bar. Using these commands, you can generate random values of a chosen parameter type for the selected row or all rows in the pattern. And, as you can imagine, these commands can lead to some interesting modulations to use as a starting point, especially when controlling automation parameters on effects plug-ins. There’s so much more to like about Step Sequencer, such as the ability to set the playback direction of rows and patterns, rotating steps within rows and patterns, loading and saving patterns and templates from the Pattern browser, and, and, and... To paraphrase Woody’s Allen’s character in Annie Hall, I love Logic’s new Step Sequencer. You know, I lo-ve the Step Sequencer. I-I love it!

Logical Intuition Logic Pro 10.5 represents a significant advance from 10.4. Jjust one of the headline new features would be a welcome part of any upgrade, but together they form a coherent workflow that doesn’t feel shoe-horned into the Logic environment (pun intended — yes, the Environment is still there). You can start in Live Loops, create regions based on clips in Tracks View, turn the results into a Quick Sampler instrument, and build a new pattern with Step Sequencer, which can then be dragged back into a Live Loops grid. Neat. There are some areas of Logic that still feel neglected, such as the Piano Roll

Redmatica Redux If some of the automatic mapping, slicing, tuning and looping features found in Sampler and Quick Sampler seem distantly familiar, it’s likely they’re based on technologies from a company called Redmatica. The brainchild of developer Andrea Gozzi, Redmatica was known for products like AutoSampler and Keymap, allowing you to create and edit EXS instruments with tools superior to those found in Logic Pro. Although there was never any official announcement Apple had acquired Redmatica, Gozzi closed up shop in June 2012 and left a message on his web site explaining he had “decided to close Redmatica to pursue other interests” before joining Apple in August of that year. Until now, the only obvious sign of this acquihire was MainStage 3.1’s inclusion of a simplified version of Auto Sampler back in 2015, the dionym becoming split across two words. Auto Sampler is now also included with Logic Pro X for the first time in version 10.5.

As its name implies, Auto Sampler is a handy plug-in for creating EXS-format instruments by automatically sampling either a software or external MIDI instrument or sound source. Simply specify how detailed you want an instrument to be sampled in terms of the note range and number of velocity layers, along with some other handy options for the capturing round robins, sustaining sounds, automatic looping and one-shot triggering, and click Sample. It’s then time to enjoy a cup of Earl Grey whilst Auto Sampler carries out its work, or even a hearty meal following by a nap depending on the number of samples to be recorded. I’ve used Auto Sampler since the Redmatica days, and more recently in MainStage, to satisfy my desire for using old Roland synth patches and it really is a terrific tool. In fact, it’s a little surprising Apple waited five years to add this plug-in to Logic Pro, although its inclusion makes sense in 10.5 alongside the new sampler plug-ins.

Auto Sampler was first reborn in MainStage 3.1, but is now part of Logic Pro X. Unlike most modern plug-in editors in Logic, Auto Sampler’s interface isn’t visually scalable.

editor and the fact that surround sound handling hasn’t evolved much since its introduction in Logic Audio 4.5. However, there are many competing products that remain largely stereo, and which offer a similar level of note-based editing. It’s only when you compare Logic with, say, Cubase Pro, that such shortcomings become more apparent. However, Cubase Pro costs £499 from Steinberg’s online shop (and requires a USB-eLicenser) with additional charges for updates, though this is perhaps an unfair comparison, given that Apple obviously don’t have to rely on sales of Logic Pro X to keep the company afloat. Logic Pro 10.5 is described in typical Apple fashion as the “biggest update to

Logic since the launch of Logic Pro X”. However, despite the inherent hyperbole of such a statement, after having worked with this latest incarnation, I’m inclined to agree. Certainly, the application has matured considerably in the seven years since the launch of Logic Pro X, and each release since has brought with it a lavish assortment of new functionality and content, while at the same time modernising a program that was originally released nearly 30 years ago. It’s probably the best £199.99 you’ll ever spend on music creation software. ££ £199.99 including VAT. WW www.apple.com

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Apogee Clearmountain’s Spaces Convolution Reverb Plug-in For most of us, Bob Clearmountain needs no introduction; he’s worked with everybody from David Bowie to the Rolling Stones. His first collaboration with audio interface manufacturers Apogee resulted in a set of plug-ins called Clearmountain’s Domain, which emulated his go-to signal chain. Clearmountain’s Spaces was originally part of that chain but it’s now available separately. Based on impulse responses, it incorporates profiles of some of Clearmountain’s favourite rooms and echo chambers, along with some of the processing used on many of his hit records. AU, AAX and VST plug-in formats are supported (Mac OS 10.12.6/ Windows 10 and higher) and protection is via an iLok account, though no physical iLok is required. It is beautifully simple to use. There are three ‘spaces’, each with its own slider and on/off button for balancing its contribution to the overall reverb character — the idea is that the plug-in is used to create a Composite Space that changes character as you adjust the mix of the three elements. There’s also a three-band EQ, as well as global pre-delay adjustment and a switchable de-esser. The centre of the GUI is dominated by a goniometer-type display that shows stereo activity. The EQ, which affects the signal before the reverb, is adjusted by dragging points

Eventide Undulator Modulated Tremolo Plug-in Eventide’s Undulator is the latest in their H9 series of plug-ins, and is available both for Mac/Windows desktop computers (all the usual plug-in formats) and for iOS devices. Authorisation is via an iLok account, to either a physical iLok or a specific computer, and two installations are allowed. Eventide are donating all the proceeds from the desktop version of the plug-in to the Equal Justice Initiative and the NAACP Legal Defense and Education Fund.

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on a curve, while the de-esser (the DS button, also pre-reverb) is either on or off. De-essing a reverb feed is a common way to stop sibilants making the reverb sound splashy, so it’s a welcome inclusion. The first space, called Apogee Studio, is a shortish and fairly warm-sounding room ambience recorded in Apogee’s in-house recording studio. Next up is Mix This, a medium-length, brighter reverb recorded in the echo chambers of Clearmountain’s own Los Angeles studio. Input

it’s actually quite versatile, and it sounds great; I found it effective on vocals and instruments as well as on drum submixes. Of course, you get the usual wet/dry control to balance the composite space’s overall contribution too, so rather than being ‘just another room reverb’ for livening up individual parts or submixes, these spaces can also be applied (in moderation) to a complete mix, to add realism and complexity without obscuring the source material; if you’re after that Tamla Motown ‘all recorded in the same room’ vibe, this plug-in will be your new best friend. I also tried adding it to one of my more ambient mixes, and it both glued the elements together and added an analogue-like smoothness. Bob Clearmountain has always appreciated the unique nature of real spaces, and the great thing about this plug-in is that it allows you to combine the best aspects of three different and musically excellent spaces that are in three geographical locations. The EQ section can make a huge difference to the reverberant sound, as the various presets ably demonstrate, many using setups that Bob used on specific records. For example, one of the included preset examples is the treatment for Keef’s guitar on the Stones’ ‘Start Me Up’. In short, if you’re looking for some fairy dust to simply make things sound better, this might be just what you need. Paul White

configuration buttons, for use when the plug-in is used in a stereo insert slot, allow this space to be configured for true stereo, stereo-to-mono and stereo crossfeed routing, the last of those reversing the left and right outputs. The longest space, Roscoe, is a rich-sounding reverb recorded in a Los Angeles studio belonging to one of Bob’s friends. Despite the apparent simplicity and surprisingly affordable price of this plug-in,

££ $49 WW https://apogeedigital.com

Undulator’s modulated tremolo started life in Eventide’s H3000 effects processor, and essentially it combines a multitap delay with feedback and detuning, alongside a multi-waveform tremolo whose depth and speed can themselves be modulated. It can be used to add richness or movement to guitar sounds, pads or samples, and the effects on offer go from subtle, textural enhancement to very obvious rhythmic modulation. If you’ve used other H9-series plug-ins, Undulator will feel familiar. There are two rows of parameter knobs, faders for the input and output levels, three virtual footswitches for bypass, Fast/Slow and

Tap tempo, and a ribbon controller that allows smooth morphing between two sets of parameter settings. The footswitches can be controlled by MIDI if required (in Logic Pro Undulator can be instantiated as a MIDI-controlled effect as well as a standard insert). To program the ribbon, you drag the white dot at the tip of any knob to the required position for the left-hand side of the ribbon, which shows a blue arc from the initial knob position to the programmed knob position. Then drag the blue dot at the opposite side of the arc to the desired position for the right side of the ribbon, and repeat for any other controls you’d like to add. These

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settings are saved as part of a user patch and the ribbon may be controlled via a MIDI mod wheel or joystick. For tempo adjustment there’s Manual, DAW sync or Off, the latter handing over control to direct tempo value entry or the tap-tempo switch, and there’s also a small Retrig button that resets the tremolo and secondary modulation LFOs to the beginning of their cycles. The top row of controls adjusts the wet/ dry Mix, modulation Speed and Depth, LFO Shape (Sine, Triangle, Peak, Random, Ramp, Square, Sample and Hold, Envelope or ADSR), Feedback and Spread, the last adding a little detuning to the delays, which are otherwise fixed. If Envelope or ADSR is selected via the Shape control, the modulation is controlled by the amplitude of the audio input, and the Speed control changes to Sensitivity. On the row below are the secondary modulation controls, which again offer a choice of waveforms or an ADSR or

IK Multimedia Z-Tone Buffer Boost & Z-Tone DI Variable Impedance Guitar Preamps

Electric guitars with passive pickups are fickle things, as their sound is affected by the load they drive (the impedance of the connecting cable and the input impedance of the amplifier or pedal to which they’re connected), and this loading not only results in HF loss but it also affects the resonant frequency of the pickups. IK Multimedia’s Z-Tone products puts the user in control of these variables, by equipping a high-quality instrument preamp with the means to adjust the input impedance to best suit the instrument. There are two related products on review here, the more compact Z-Tone Buffer Boost and the larger Z-Tone DI. Taking the Buffer Boost first, this takes the form of a dual footswitch stompbox. It features a transformer-balanced, 600Ω XLR output, with ground-lift switch, for feeding mixers or audio interfaces. There’s also a buffered, unbalanced quarter-inch output jack with a 56Ω impedance. A Link jack is wired in parallel with the guitar input jack, and this can be used to feed the signal on to an amp with the benefit of the variable impedance. The Z-Tone Buffer Boost can be powered from a 9V battery, a 9V pedal power supply or 48V phantom power; phantom takes precedence over the battery when present.

envelope follower to modulate the way the main LFO behaves. Here, you can set how much the secondary modulation affects the Speed and Depth of the main LFO. A minor frustration with this plug-in is that the delay isn’t adjustable in level, so it hovers in the background of every patch; you can’t set up a ‘nothing-but-tremolo’ effect. That aside, the combination of the delay with a little feedback and detuning, plus tremolo, produces a really sweet enhancement for string patches and the like. The secondary modulation facilities make it possible to set up a tremolo that changes speed according to LFO or envelope settings, while the choice of waveforms offers a wide palette, ranging from smooth, sine-wave tremolo to choppy square or sample and hold

Its preamp can be switched from JFET to Pure. The JFET mode, which uses a discrete junction gate field-effect transistor, adds a little harmonic colour, whereas the Pure mode sounds very clean. The preamp can be switched to suit passive or active pickups too — the latter have a lower impedance than passive ones and aren’t affected by load impedance in the same way. In Active mode, the Z-Tone control, which adjusts the impedance between 1MΩ and 2.2kΩ, is bypassed and a fixed 20kΩ impedance is employed instead. There are just three knobs to adjust: Gain, which works on the input; the Z-Tone Sharp/Bold knob, which controls the impedance; and Boost, which can add up to 10dB of gain. The leftmost footswitch is Bypass, and the rightmost one brings in the Boost. Both have status LEDs. Conceptually similar, but presented in a DI-box format without the footswitches or boost option, the Z-Tone DI has a choice of battery or phantom power operation, and

steps. Given its low cost, there’s much to commend this plug-in. Paul White ££ Mac/Windows version $19. iOS version $7.99.

WW www.eventideaudio.com

has the same core control and connectivity options as the Buffer Boost. Additionally, it has a switchable 20dB pad that also bypasses the Z-Tone variable impedance control when in use, for when you need a more conventional DI box to use with higher-level sources. Both units offer a frequency response that’s flat to within 1dB from 5Hz to 30kHz.

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The dynamic Range is quoted as better than 114dBA for both outputs. I tested both units with guitar and bass, with interesting results in both cases. At the full 1MΩ end of the impedance control, the sound was much the same as going into the high-impedance input of a typical audio interface, which is a sound that I often find slightly thin and lacking in weight. Go to the other extreme and you lose about half your level, and the result sounds quite dull;

it won’t be your go-to setting but may be useful in some situations. The Goldilocks zone lies somewhere between the two extremes, and when I found it the bass sound warmed up slightly and my DI’d Strat sounded just that bit more solid, with a less ‘glassy’ high end. The choice of Pure or JFET modes brings about a subtle difference in tone, but it’s useful one. I generally preferred the more ‘organic’ sound of the JFET option.

Soundevice Digital SubBass Doctor 808

leaves that large and enigmatic Cure knob in the middle of the window. The manual says rather vaguely that it provides ‘Frequency solving of your bass’, so I had to resort again to a spectrum analyser. From what I could see, it shapes the added sub-bass component and reduces the level of extremely low (sub-sub?) frequencies that might otherwise induce a panic attack in your speakers. A call to the designer confirmed that it helps tame what’s going on below 40Hz, but that it also has some effect on the harmonic structure. The ‘how does it sound?’ question rather depends on the frequency range of your listening system, as the synthesized frequencies that are added will be below the range of some smaller speakers; you may have to resort to headphones and meters to find out what’s really going on. Unless the sound being processed is pretty deep-sounding to begin with, little or nothing seems to be added, so this really is a treatment for kicks and bass instruments. Those caveats aside, though, SubBass Doctor 808 produces very smooth-sounding, natural results, and I really like the way the saturation section helps to beef things up without

Sub-bass Generator Plug-in

As its name might suggest, SubBass Doctor 808 is particularly well suited to EDM drum processing but it has applications in other genres too. It supports the common Mac and Windows formats, and Authorisation is via a personalised key file, so you can install it on all your machines. SubBass Doctor 808 is designed to add sub-bass to sounds that already reside in the low-frequency register. As the information provided with the plug-in doesn’t say what frequencies are generated, I checked with a spectrum analyser, which showed that almost all the added information was sub-50Hz, which is sub indeed! The resizable GUI sports five knobs, plus metering for the input and output, but as two of the knobs set the input and output levels, there are only three that are used to tweak the effect itself: Sat%, Cure and Sub. Sat% adds saturation, but only to very low frequencies, so its effect is more to compress and enrich rather than to add obvious dirt. It works in parallel with the sub generator. Sub simply regulates the level of the sub component that’s being added back into the signal path, which

Neunaber Wet Reverberator Reverb Plug-in Just when I thought the last thing the world needed was yet another reverb plug-in, Neunaber came along with something a little different. Neunaber are best known for their high-end guitar reverb effects pedals, but they’ve now ported their reverb algorithm into software, to create the Wet Reverberator plug-in (VST2.4, VST3, AU and AAX formats) for Mac and Windows. Authorisation is via a license key code, which removes the time restriction from the otherwise fully functional demo . Neunaber tell us that Wet Reverberator is based on an algorithm that has been refined over the last 10 years and is

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As a solution to fine-tuning guitar and bass sounds prior to DI’ing, both units do an effective job, and both are very solidly built too. In a live performance situation, the pedal version would be more practical, as it allows both for bypassing and adding boost. But either would do a fine job in the studio. Paul White ££ Z-Tone Buffer Boost €207.39. Z-Tone DI €182.99. Prices include VAT.

WW www.ikmultimedia.com

compromising the overall sense of clarity in the sound. By combining the Sat and Sub processes, you can add serious weight to bass sounds and, unlike some other sub processors I’ve tried, the control range prevents you from adding too much or processing frequencies that are best left alone. That being the case, it’s pretty difficult to make this plug-in sound bad and those creating EDM for reproduction on club sound systems will find a lot to like here. The plug-in comes with a set of presets, of course, but with so few controls you barely need them. Paul White ££ £52 including VAT. WW https://unitedplugins.com

used in their Immerse Reverberator MkII pedal. The plug-in, though, has an expanded feature set. Currently (v1) the plug-in is stereo-in, stereo-out only. So if you want to use it on a mono source in Logic Pro X (the DAW I used to test it), it’s necessary to set the track to stereo and to insert a Direction Mixer plug-in before Wet Reverberator, with the width set to zero to place the dry sound in the centre; otherwise the dry sound only comes out of one channel. I’m told this will be resolved in a future update. The GUI is clearly laid out, with slider controls for the values and a very useful Visualiser in the middle of the screen that shows both the current EQ settings and the reverb’s density as it decays. To the left of the screen under the Size and Shape heading are four faders

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ON TEST

which control Mix, Delay (pre-delay), Attack and Decay. Attack smooths out the onset of reverb to soften transients such as heard in picked guitar or drums. Decay covers an enormous range from room ambience to near-infinite cavern-like drones. EQ is provided via sliding first-order low-cut and high-cut filters, and there’s also an EQ tilt control that pivots around 500Hz. Below the Visualiser display there are two further sliders for setting the damping frequency and attenuation to emulate the way reverb decays in a real space. To the right is a modulator section and a Fidelity section where you can apply bit-reduction and control the stereo width from mono to extra wide. The LFO-based modulation has controls for Rate and Pitch, and while it isn’t possible to turn the modulation off, the effect of modulation can be set to a minimum by setting the Modulation Controls to ‘LFO: 1’, with rate and pitch at their minimum settings. There’s a choice of two LFO configurations

Sugar Bytes Looperator For iPad Step-based Multi-effects Processor Sugar Bytes have acquired a loyal following for their undoubtedly leftfield, quirky and sometimes just a bit bonkers line of desktop plug-ins. To their credit they’re one of few desktop developers who have fully embraced the mobile music-making environment, and they’ve brought many of their popular desktop plug-ins to iOS. The latest to undergo this journey is Looperator, a ‘free to try’ download from the App Store, whose full feature set can be unlocked if you like what you see/hear via an in-app purchase. The app requires an iPad running iOS 12 or later, and works both standalone and as an AUv3 plug-in. MIDI Learn, Ableton Link and cloud-based preset sharing are also supported. As far as I can see, the iPad version of Looperator is pretty much a complete port of the desktop version I reviewed in SOS November 2015 (https://sosm. ag/sugar-bytes-looperator). To recap the underlying concept, you feed Looperator an audio source, which it will then slice into 16 steps. For each step, you can apply a series of effects that include filters, stutters, slices, distortion, tape stops, volume modulation and looping. Based on whatever step length you specify, the pattern of step-based effects is then cycled in sync with the tempo (of the standalone app or your AUv3 host) while playback of the audio source (live or prerecorded)

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which affect the way it works on signals within the reverb algorithms and it’s most evident at long decay times; you just pick which works best. Sonically, Wet Reverberator is a gift to anyone into ambient music, as it is possible to create lush washes that don’t overpower the dry sound. But at shorter decay settings it also fares well in most conventional applications. It’s difficult to quantify just what makes a reverb sound good, but this one is a bit special. I don’t often use bit reduction, but it can be useful here if you want to add a bit of dirt to short snare reverbs and the like. I hope that at some point Neunaber also decide to port their Shimmer Reverb algorithms to a plug-in format, but in the meantime you can get pretty close to that effect by using Wet Reverberator in an aux send, and putting an octave-up pitch-shifter set to a 50-percent wet/dry mix in front of it. I’m sure this will become one of my go-to reverbs. Paul White ££ £38.70 including VAT. WW https://neunaber.net

continues. It is, frankly, super cool, and particularly good for creating rhythmic effects to spice up your latest electronic music mix. As on the desktop version, the main display shows the waveform of your incoming audio and the six available ‘lanes’ of effects that can be applied. Each lane focuses on a specific collection of effects and, at any step, your audio is processed from top-to-bottom in the effects chain. Lanes can be reordered, though, and you can also adjust the wet/dry balance in various ways. There’s an impressive range of general presets to get you started, as well as presets for each of the various effects types. However, you can also define your own user effects in each lane. As this seems to be a pretty complete port, interested users might want to dip into the SOS archives and read the more in-depth desktop review, but the bottom line is that Looperator provides a huge range of creative effects options. While the functionality of the plug-in mirrors that of the desktop, the experience of using it on an iPad under iOS is rather different, of course. First, I experienced absolutely no problems using the app either standalone or as an AUv3 plug-in hosted by

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Cubasis 3.1. The UI is very nicely designed, and there are plenty of things to touch; if your AUv3 host allows you to switch it to a full-screen display as you tweak (as Cubasis does), exploring Looperator’s many creative options is a much more satisfying experience. Beyond that, I can only repeat what I said when reviewing the desktop version. While Looperator can do subtle if required, it excels in those attention-grabbing special effects that provide a production-style sonic hook for your track. In this role, it’s a heck of a lot of fun. It might not appeal so much to singer-songwriter types, but if your musical muse lies in any form of electronic music, and you like to spice up your mixes with some creative effects, Looperator is well worth exploring. John Walden ££ Free to try; £17.99 in-app purchase to unlock full feature set.

WW https://sugar-bytes.de

VIDEO DOCUMENTARY ORIGINAL S IN ASSOCIATION WITH

Pete Cannon: Producing Jungle The sound of jungle is intrinsically linked with the computers and samplers of the ‘80s and ‘90s. In the latest SOS Video Feature, we talk to jungle producer Pete Cannon about how and why he’s still using Akais, Ataris and Amigas.

www.youtube.com/soundonsoundvideo

ON TEST

Sample Logic

Cinematic Guitars Motion

SAMPLE LIBRARIES

Kontakt Instrument

HHHHH Sample Logic’s Cinematic Guitars series is a well‑established part of many a media composer’s sonic arsenal. Built around sample sets derived pretty much exclusively from guitar sources, the front end combines ‘quad core’ sample playback (four sample engines can be combined in a single preset), sophisticated effects for each core and a master effects chain, step-based control of both effects and pattern/arpeggiation plus an XY pad to create additional sonic motion, and that can also be animated. Shipping with massive collections of presets, and tools for randomising sound creations so something new can always sit under your fingertips, they have supplied many a cue for films and TV shows. The latest iteration of the franchise is Cinematic Guitars Motion and users of the earlier titles (such as Cinematic Guitars Infinity, reviewed in the December 2015 issue of SOS) will find lots that’s familiar in terms of workflow. However, this time around, Sample Logic have brought the UI bang up to date, made some changes under the hood, enhanced various aspects of the feature set and, of course, built the latest title from a brand‑new 33GB library of samples. Requiring a full version of Kontakt 5.8.1 or later, it has to be said that CGM is an impressive beast. With over 850 instruments and presets included, there is plenty to explore even before you dig into the multitude of possibilities offered by the powerful front end. The ‘motion’ part of the title is apt as many of these presets make very good use of the step-based pattern features, the effects and the XY controller that can adjust the blend between the four sound cores in real time. And, while I was perhaps a bit surprised that the new and improved preset browser didn’t include a ‘rhythmic’ tag amongst its other useful descriptors, as so many of the preset sounds offer rhythmic elements — either via note arpeggiation or step-based effects — CGM can easily set the pace of a cue as well as create its harmonic/melodic properties.

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There are lots of presets here that could easily generate a complete short cue without the need for any other sound sources. Whether used for tension, action, drama or more melancholic and calm moods, CGM is a powerful tool once you dig in. Perhaps the only other thing to note is that, while there are plenty of organic types of sounds included, this is perhaps more of an electronic or synthetic sound source. It could, however, easily be combined with more traditional orchestral‑type sounds. As with earlier Cinematic Guitars titles, pricing suggests this is targeted at the working media composer and, if that’s you, and deadlines are always short, then Cinematic Guitars Motion will most certainly warrant serious consideration. Hefty in size and price but matched by some equally hefty sonic possibilities. John Walden $349.99 www.samplelogic.com

Orchestral Tools

Tableau Solo Strings Sine Player

HHHH This tasteful library is the flagship of the Organic Samples range, created by young composer Maxime Luft in conjunction with the Orchestral Tools samplemeisters. Tableau Solo Strings (60GB) consists of violin, viola and cello solo instruments recorded at the Teldex Scoring Stage, Berlin, from five mic positions. The same hall and miking scheme are used in Orchestral Tools’ other collections, so these solo strings should fit nicely into the family. The instruments may be bought separately, or together as a discounted bundle. The violin samples were performed by Farida Rustamova on a 1745 Guarneri instrument. Despite its antiquity, the old fiddle sounds full of life. Its rich, vibrant, singing tone is ideally suited to melodic playing, and its swooping portamento slides are a joy. Long notes are played in a variety of attacks, with and without vibrato. You can’t crossfade between the two, which is a pity. While the player’s strong, expressive vibrato style is great for stirring romantic lead

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lines and dramatic themes, I found it a touch overbearing for simple folk melodies. This solo cello’s liquid-sounding legatos and soaring portamentos are a pleasure to play, and its reflective, noble tone is a great melodic resource. Cellist Vasily Bystroff also turns in some brilliant, biting and visceral short notes; when he digs into his louder spiccato deliveries, you can almost see the rosin flying off the bow. In an unusual twist, the cello’s trills and tremolos start slowly and gather pace, adding a mobile expressive element to standard articulations. My ‘best in show’ award goes to the viola. Played by Alejandro Regueira, this superb instrument’s short spiccatos range from light, insistent brushes to a furiously emphatic bowing, a great articulation for driving ostinatos. The viola’s dark, mellow tone also works a treat for stately heroic themes and slow-moving ballads; its legatos are less adaptable to fast playing, but if you’re prepared to massage your MIDI notes after the event, they can produce good melodic results. On the downside, the note attacks of the non-vibrato articulations are generally slower than those of the vibrato samples, which can be a problem when combining both in a melody. All instruments utilise a superior VSL‑style legato mode, and play sustains, marcatos, pizzicatos, tremolos, semitone trills and four types of mordent grace note. These solo strings are beautifully recorded and played with great commitment, and the samples successfully capture the personality of each player. On a technical note, the library runs on Orchestral Tools’ free Sine

Player, which connects to the company’s online store and allows you to download instruments and mic positions, rather than waiting for the whole shooting match to trickle down the wire. It’s a great innovation which allows buyers to get straight to work with these classy samples. Dave Stewart €119 www.orchestraltools.com

Sonic Atoms

Baltic Shimmers Omnisphere Expansion

HHHH Originally developed for use with the free Halion Sonic SE 3.4 sample player, Baltic Shimmers is now also available as an expansion pack for Spectrasonics Omnisphere and is described by its Polish creators as a collection of ambient sounds inspired by the sea. Sound sources include the sea, birds, acoustic and electronic musical instruments, and a lot of processing, which apparently includes recording many of the sounds to tape. While the sounds were created with film and game scoring in mind, they also work well on minimalist music. The instruments are categorised as Shimmers, Pads and Drones, each with its own set of presets. In all there are 144 presets and the pack includes around half a Gigabyte of samples. Sounds in the Shimmer category can be made to morph in three stages from a fairly static sound into something more motional, which the designers suggest is the equivalent of going from a clean sea to waves or foam. Some work as evolving pads while others are better played as single notes. Sounds in the Drones section can vary in level, timbre and sometimes pitch as they evolve, so they are best suited to single‑note work. These vary from tranquil to mildly disturbing. Drones work best played monophonically and tend to evolve over a long period, which can be extended by applying modulation. Pad sounds are made up from three layers, the first two of which are long, textural sounds played either forwards or backwards, while the third is a sea type of sound triggered when the key is released. I found that Omnisphere’s Orb provides an easy way to add further dynamic variation to the sounds. The mod wheel generally affects patches by changing their level or character. Some of the patches have an element of cloudiness about them while others pulsate and flow. Many of the sounds make ideal beds for minimalist music with the long, evolving sequences keeping things interesting without

being too attention grabbing. If you make relaxation music, then head for the Pads section — just add a reverb‑soaked piano melody and you’re home. All the sounds have an organic quality about them. You wouldn’t generally use these types of sound to play melodies, but the pads work well for sparsely voiced chords. You can also create your own sounds by taking layers from the different presets and recombining them so there’s plenty of scope to push the boundaries if you don’t mind a little light editing. Similarly, changing Omnisphere’s filter settings, using the Orb or changing effects helps ring the changes. Given its low cost and the high quality of the sounds on offer, Baltic Shimmers is perfect for its original game and film design brief but it also has a role to play in experimental, chillout and minimalist ambient music composition. Paul White £34.99 www.timespace.com

Ergo Kukke

Trails

Kontakt Instrument

HHHHH Designed by musician/composer Ergo Kukke, and available via Kontakt Hub, Trails is perhaps best described as a blend of playable instrument, rhythm/pulse creation and abstract sound design. The library is primarily aimed at media composers, so if ‘epic cinematic composition meets sound design’ is your thing, Trails has the potential to appeal. Sonically, Trails is built from nearly 13GB of sampled sounds and ships with a powerful, custom-built Kontakt (v.5.8.1 or later required) front end. The 270+ Snapshot presets are divided into five categories — Atmospheres, FX, Menus, Oneshots and Rhythmic — although each uses the same UI. Like a number of other current composer-targeted tools, Trails features a multicore sound engine, allowing you to blend up

to four of the underlying samples. The interface is laid out in a very logical fashion, with the four sound engines arranged around a central XY pad that allows you to adjust the blend between them, either manually or via automation configured on the Movement Page. Other pages of controls provide access to the comprehensive effects and modulation options. Within the Main page, each of the four sound engines are identical. All the usual features are present, including a drop-down menu for sound selection, basic mixing/tuning features, an ADSR envelope, multimode filter and three‑band EQ. It’s straightforward in operation but, by the time you combine all these options, you have some mightily impressive sound‑design possibilities. If you need some help, Trails includes a randomisation system that operates at multiple levels. This is actually very well implemented. Of course, this would all be so much fluff if the sounds themselves were anticlimactic; that’s absolutely not the case as, once you dig in, Trails sounds fabulous. Given the five different categories of sounds, a few instances of Trails is really all you need to create your next epic cinematic music/sound-design crossover cue. The potential for pads/atmospheres is huge, the sound effects and one-shots make it easy to add tension and resolution, and there is enough melodic potential to make it genuinely musical (as opposed to pure sound design). For drama, sci-fi, horror or otherworldly atmospheres, Trails provides a very creative toolset. Perhaps the only downside is the price, and while that might be beyond some, it’s still very competitive given the products it is competing with. If you want a cinematic sound-design/ compositional tool that does dark, moody, impactful and atmospheric, and are prepared to look beyond the usual suspects for something a little different, then Trails Cinematic Visions is most certainly worth auditioning. John Walden £174.50 www.kontakthub.com Audio examples of this month’s libraries are available at www.soundonsound.com.

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T E C H N I Q U E / D I G I TA L P E R F O R M E R

Beats Of Burden The latest version of DP includes some powerful timing & tempo-manipulation tools. MIKE LEVINE

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igital Performer has long had excellent features for manipulating beats and tempo. With the arrival of DP10, those capabilities have been significantly upgraded. This month we’ll look at some tips for using DP’s beat‑detection, time-stretching and tempo-related features. The new Stretch Edit Layer, which is available in both the Sequence Editor and Waveform Editor, takes advantage of the new Beat Detection Engine introduced in DP10 to offer additional functionality for editing beats and changing tempos. You can turn on Stretch functionality for an audio track by going to its Track Settings Menu in the Sequence Editor and selecting Stretch. Even if the audio you’re importing doesn’t have embedded tempo information, DP will automatically conform the audio in the track to fit the project tempo. This feature will work whether you’re using the Tempo Slider or the Conductor Track. The most obvious use for setting a track to Stretch is to bring in sounds from outside the project, such as loops, The Beats layer in DP10’s Waveform Editor.

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and conform them to the project tempo. If you Stretch-enable all the audio tracks in a project, you can freely change the project tempo; all the tracks will automatically adjust along with the changes. Stretch will even follow multiple tempo changes in the conductor track. The quality of the ZTX Pro time-stretching technology that DP employs is excellent and flexible, so it can be applied to all types of audio material. If you know in advance that you’re going to be experimenting with song tempo, it’s easiest to go to Preferences / Pitch and Stretch, and check Stretch Enable in the Default Options for Tracks in New Projects

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When Stretch is enabled on audio tracks, you can add tempo changes in the Conductor Track, such as the ritardando at the end of this song.

and Default Options for Tracks in Current Project categories. That way, every track you create will be Stretch Enabled from the start.

Tempo Changes You can use the Stretch capabilities of audio tracks, in conjunction with the Conductor Track, to program in tempo changes, such as a ritardando or accelerando, or to just add some slight tempo variations into a song recorded to a click. This gives you a lot of arrangement flexibility in a mix or remix situation. To do this, first check that all your audio tracks are set to Stretch in the Track Settings Menu. Next, make sure your tempo is being controlled by the Conductor Track. If not, make a note of the song’s bpm setting, and hold down the disclosure arrow to the right of the Tempo Slider and change it to the Conductor Track. DP’s tempo will change to the default of 120. Open the Conductor Track in the Event List and put in a tempo change at bar 1:1:000. If you don’t remember what the original tempo was, you can quickly switch back to the Tempo Slider to

see and then return to the Conductor Track and enter it in the Event List. Next, find the spot in the song where you want to change the tempo. Say you want to add a ritardando in the last measure, ending on the last beat. Go to Project / Conductor Track / Change Tempo, and use the Change Tempo dialogue box to program the change in. Alternatively, you can use the Pencil Tool and draw it into the conductor track. Hit Play and you’ll hear all the audio and MIDI tracks follow the tempo change. Adjust as needed.

A Stretch In Time You can also manually stretch beats to correct or change a part, either in the Waveform Editor, which has been reorganised and spruced up in DP10, or in the Sequence Editor. The process works best for small corrections. Once you start dragging beats too far, you’ll notice degradation of their transients and impact the timing of surrounding notes. One thing that helps is to experiment with different Snap-to-Grid settings when you’re making the adjustments. The larger the note value in the grid, the more any stretches you make will impact surrounding beats. Beats are bordered on either side by a red line that represents their Anchor Points. You can move these around if you need more control over what gets stretched on either side of the beat. Hover over the line until you see a bi-directional arrow, and then drag in either direction. Another option for time-correcting beats is to use the Region / Quantize / Beats Within Soundbites command, which allows you to quantise both audio and MIDI material together in one operation. Just like with quantising MIDI, the key to successful results is to set the parameters such as Grid Value, Sensitivity and Strength, so that you get just enough time correction without removing the feel.

Using the Stretch Edit Layer, you can manually drag audio in time to make changes or corrections to the rhythm.

From there, you could use the Quantize command set to Soundbites to correct or change the part. Or you could drag the newly separated Soundbites manually to where you want them. Another reason to make Soundbites from the beats is to create samples to load in a sampler instrument such as DP’s Model 12. You can also switch the Waveform editor to show beats, and edit them if needed, before separating a Soundbite by its beats or using another beat operation. You can even raise and lower beat velocity, which will

impact which beats are detected when you increase the threshold. Drag the handles on the beats down to reduce their velocity. You can also add beats with the Pencil Tool or delete them with the Mute Tool. Remember that, unlike the Stretch layer in the Waveform Editor (or Sequence Editor), editing beats doesn’t alter the audio directly. What you’re editing is a map of the positions of the beats in a Soundbite, which DP will use as reference when performing beat and tempo operations.

MIDI Beats Another trick the Beat Detection Engine can perform is to covert beats into MIDI

Bite Me An alternative method for manipulating beats is to use the Audio / Audio Beats / New Soundbites option from the Beats menu to create new Soundbites based on the audio beats in the original. These new Soundbites will start precisely on their initial transients. When you select New Soundbites, you’ll get a dialogue box with a Beats Strength Threshold slider, which will adjust the sensitivity of the beat detection. If too many beats are being detected, push the slider more to the right.

Using the New Soundbites from Beats command, a single Soundbite has been cut into many smaller ones, all starting right on the beat.

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information. To do this, select the Soundbite with the beats you want to convert and go to Audio / Audio Beats / Copy Beats as MIDI. You could use this function in several different ways. Say you’ve got an audio percussion track that has a nice groove to it, but the sound or instrument is wrong. You could convert it to MIDI and use a sample of a different percussion instrument. Bear in mind, though, that because beats have no pitch information, the MIDI track you create with the Copy Beats as MIDI command will all be on one note: C3. You’ll almost certainly have to go in and do some editing of the pitches to get them to trigger the desired sound. You could even use it on an audio drum loop that you want to convert to MIDI. The transposition would be time-consuming for a song-length part, but if you’re just converting a loop of a few bars, it only takes a minute or two.

Adding A Click You can also tap into the power of DP10’s new Beat Detection Engine to create a click track and grid for a piece of music recorded with no tempo reference. The process works best on music with a prominent tempo and without rubato sections. For this operation, open a new sequence, and import the mix onto a new track. If the music in your file has a pickup, so that the first hit isn’t on beat 1, you’re best off temporarily cutting the Soundbite to remove the pickup. In order to make the tempo-detection process line up with the measures in the timeline, the downbeat of the first full measure of music needs to start at the beginning of a bar. Cut right on the line indicating the detected beat, then place it at the beginning of the measure where you want it (typically 2:1:000). Make sure that it’s precisely located by opening the Event List for that track and, if necessary, typing in the correct location. Next, select Audio / Audio Tempo / Analyze Audio Tempo. Then choose Audio / Audio Tempo / Adjust Sequence to

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The MIDI notes in the track here were extracted from the drum track above them using the Copy Beats as MIDI command.

Soundbite Tempo. If you don’t have the tempo set on the Conductor Track, you’ll get a dialogue box asking if you want to switch to the Conductor Track or stay with the Tempo Slider and use an average setting. Choose the Conductor Track option for the most accurate results. Listen with the click turned on to see how well things are lining up. DP’s Beat Detection Engine is smart enough to be able to figure out a tempo from beats even when it’s showing intricate The tempos on the right were calculated by DP to create a tempo grid for this song, which was recorded without a click.

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patterns of transients. Just because you see a considerable number of beats inside a measure, that won’t necessarily create problems for tempo analysis. That said, there are some songs where you’ll have to do some additional editing. You might need to insert meter changes to match any that may be in your song. You also might find that DP analysed your song’s tempo correctly but made the click twice as fast or twice as slow as you want a quarter-note value to be. In those situations, try selecting Audio / Audio Tempo / Halve Soundbite Tempo, or Audio / Audio Tempo / Double Soundbite Tempo. You can do further manual adjustments, if necessary, using DP’s Adjust Beats feature, which is accessed via Conductor Track / Adjust Beats.

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TECHNIQUE / REASON

External Influences Using MIDI controllers with Reason.

Screen 1: The Control Surfaces preferences is where MIDI controllers are usually set up.

mapping schemes (‘Remote Codecs’) that Reason uses for this purpose. This is in line with how most DAWs work, but alongside this Reason has a separate direct MIDI routing system specifically for patching incoming MIDI channels directly to devices in the Rack. This method is great for connecting an external sequencer, or for using the Reason Rack purely as a multitimbral sound module.

MIDI Patchbay SIMON SHERBOURNE

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’m increasingly turning to hardware sequencers to make music, but even when I’m not using Reason as a DAW or plug-in it’s still my favourite sound module. Reason has two completely different systems for configuring MIDI sources. This month we unpick some of the confusion around this to find the best ways to sequence and play Reason from external devices.

Incoming Most Reason users connect a keyboard controller to play Reason’s instruments, tweak controls and record MIDI parts into

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My KeyStep Pro sequencing Rack, with a drum machine and three synths.

the Sequencer. Unless told otherwise, Reason assumes that a connected device should act like a master, and will direct all MIDI from it (regardless of channel) to whichever track or device is selected. You’ll find the device/port listed in Reason’s Control Surfaces Preferences (screen 1) under ‘Easy MIDI Inputs’. Reason has an intelligent MIDI control and mapping system called Remote. Controllers that you declare for Remote use (also in the Control Surfaces Preferences) can take advantage of fluid mapping of controls to Reason’s devices. Many MIDI controllers have predefined

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The direct MIDI routing system lets you elect up to four MIDI ports to appear on a virtual MIDI interface at the top of the Reason Rack. These are chosen in the Sync tab of Reason’s Preferences, under the External Control heading (screen 2). Simply choose MIDI ports from the pop-up lists next to each the four buses. Next, go to the Rack, scroll to the Hardware Interface at the very top, and show the Advanced MIDI panel (screen 3). The panel has four buttons for viewing the busses (MIDI Ports), with panes for the individual channels. The pop-ups list every device in the Rack, and set the routing target for each incoming MIDI channel. Note that Mix Channels modules are shown as well as instruments

Screen 3: The Advanced MIDI panel of the Rack’s Hardware Interface lets you patch incoming MIDI channels to devices.

KeyStep’s Drum track on channel 10. The other three synths are played by the three Arp/Sequencer tracks available on the keyboard.

Note On Sync

Screen 2: Ports you want to appear in the MIDI Rack Interface are selected in the External Control section of the Sync Preferences.

and effects (so that you can automate them from external CC sources). There’s often some overlap in device names, so be careful that you’re routing to the right place. The main screen shows the Rack I’ve set up for sequencing with an Arturia KeyStep Pro, which has four channels. A Kong drum machine is patched to the

When you’re using Reason purely as a sound module, you might not need to sync Reason’s transport. However, you might want to send it MIDI clock from an external source so that synchronised effects and modulators match your external sequencer’s tempo. You can choose your MIDI clock source in the Sync Preferences, below where you set up your control busses. However, while you might not need to think about timeline sync, you do need to be aware of audio latency when using Reason as an external sound source. The audio buffer (in Audio Preferencess) determines how quickly you’ll hear a sound from Reason after it’s triggered from your external sequencer

Lock Box Using Reason’s External Control system is not the only way to directly patch MIDI to a Rack device. MIDI controllers (or other MIDI source ports) that are enabled in the Control Surfaces Preferences can be Locked to a specific device, either from the Edit menu or right-click menu (screen 4). The port will subsequently only communicate with that device, regardless of Rack or track selection in Reason. In fact, there’s a couple of notable advantages to working this way. Firstly, you can take advantage of the MIDI control learning, in other words use the Remote Override Mapping feature to control parameters without having to look up and configure CC values on your hardware. Secondly, you can record into Reason tracks.

Screen 4: As an alternative to using the Advanced MIDI Hardware Interface, single‑channel sources can be locked to a specific device from the right-click menu.

MIDI routed via the External Control busses has no connection to the Sequencer, which is a problem if you hoped to capture your external sequences as MIDI in Reason. So why not go this route all the time? The answer comes down to MIDI Channels. Connections via Remote and the Control Surface setup page are port-based. So you can direct or lock any MIDI port to a specific device, but can’t specify a channel. The External Control interface splits out each port to multiple channels and makes them available in the Rack. The example I’ve used here with the KeyStep Pro would not be possible without using the Advanced MIDI interface, as the four sequencer channels are MIDI channels within a single port. Incidentally, I could set up the KeyStep Pro as both a Remote-style master keyboard and an External Control sequencer, but it would require connecting it via two separate ports: the USB, and a standard MIDI port via an interface. I suspect I could set up some clever filtering by connecting via a Retrokits host like their new RK-006. With more controllers offering both master keyboard and sequencing, it would be great if Reason could offer channel selection as part of its Control Surface setup.

or controller. If this delay is a noticeable amount, sounds triggered in Reason will be out of time with other stuff you’re sequencing. The simplest way to fix this is to set the latency of your audio driver as low as possible (I try to never go above 128 samples). Your hardware sequencer may also have some way to offset MIDI timing to different outputs, but this doesn’t help with live playing.

Continuous Control As well as note sequencing you can use the External MIDI Bus connections for controlling and automating Rack devices from knobs, sliders and other MIDI CC sources. However, because patching via the Rack interface is not part of the Remote system, you can’t simply map (learn) any MIDI source to a device control with the usual Remote Override option. Instead, you’ll need to use the predefined MIDI CC assignments built into the Rack devices. You can find these in the MIDI Implementation Chart that’s part of Reason’s documentation. For example, Filter A on the Malstrom synth in my Rack responds to MIDI CC#79. The KeyStep Pro has a dedicated controller bank of five encoders. I configured the first of these to send on controller 79, and I set its MIDI channel to 4 to match the note track patched to the Malstrom. Now I can both sequence and automate the synth from my hardware.

Inter-app Traditionally, the External Control busses have also been used to trigger Reason from other software apps running on the same computer, especially if they didn’t support Rewire. The need for this has faded with Reason becoming a plug-in: it’s easier to run Reason’s instruments directly inside your DAW than to set up internal MIDI busses. However, if you prefer to use Reason as a complex multi-device Rack environment, the original method may still appeal, as the Reason plug-in can’t accept multiple MIDI inputs. Macs have an internal MIDI port available by default, which you can see as IAC Driver Bus 1 in the screens. For Windows users I’d recommend installing loopMIDI.

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TECHNIQUE / PRO TOOLS

Split & Polish We explore Avid’s Pro Multiband Splitter plug-in. JULIAN RODGERS

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’ve always felt the Pro Series plug-ins from Avid are unsung heroes of the plug-in world. They’re provided to subscribers as part of the Pro Tools bundle, so they are almost ubiquitous, and they provide a definite step up in terms of features and quality compared to the stock plug-ins. I’ve spoken about some of these plug-ins in the past in this column, but one of the most useful is the Pro Multiband Splitter. It’s a simple plug-in — basically the same as the Pro Multiband Dynamics processor, but without the dynamic processing. One thing that many people don’t realise about the Pro Multiband Splitter is that you can output each band to a bus and, via that bus, to anywhere else in the mixer. These busses are automatically created and are accessed via the input selector in the plug-in menu. The Multiband Splitter gives access to these splits and nothing else.

The Small Phases An important point to make about both the Pro Multiband Dynamics and the Pro Multiband Splitter is that the filters used in their crossovers are of the minimum‑phase variety. Linear‑phase filters have been available for a long time — the first example of a phase‑linear multiband compressor I ever used was the Waves Linear Phase Multiband Compressor back in the early 2000s — but the issue with linear‑phase filters has always been that to

Avid’s Pro Multiband Splitter plug-in lets you separate audio into different frequency bands and send those bands separately to the mixer.

exhibit the phase response that they do, they necessarily incur significant latency. For the majority of applications the sonic differences between a minimum‑phase and a linear‑phase filter is subtle, but the latency of a linear‑phase filter is large, so in most cases, minimum-phase filters are a perfectly acceptable compromise. That being said, the provision of a choice between minimum phase and linear phase filters would be a nice option to have. The FabFilter Pro MB is one of several plug-ins that come to mind that offer such a choice, but unfortunately it doesn’t offer the option to split each band out separately in Pro Tools.

Loudness Analyser Another unsung ‘utility’ plug-in in the Pro Series is the AudioSuite Loudness Analyser, which complements the Pro Limiter very well. For anyone working to loudness standards this is a very straightforward problem solver. Since the popular music streaming platforms started using loudness normalisation, LUFS are no longer only relevant to those working in post production and broadcast. If you distribute music online then understanding loudness is something you need to think about. The Loudness Analyser offers an offline, faster‑than‑real-time measurement of integrated loudness, as well as peak and loudness range measurements. While more comprehensive third‑party tools are available, as someone who works to loudness standards every day I have to say this is all I need.

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The significance of the phase response of the filters is that the crossover in Pro Multiband Splitter isn’t completely transparent, as can be demonstrated if you try to null it against a polarity-inverted parallel path. For most purposes this doesn’t matter, but for processing entire mixes it is something to be aware of. It’s not necessarily a bad thing but, particularly if it is being used in a parallel application, it will colour the sound just by being there, even before any processing has taken place.

Getting Creative So the concept is clear, but why would you want access to these split bands, and are there any caveats to be aware of while using them? The idea of inserting a flanger on the low mids, a reverb on the top end and Auto-Tune on the sub-bass might be original but it’s unlikely to be useful. How could this ability to route and process each band separately be used to solve issues we actually experience when mixing? I won’t refer to any compression examples as these might be better served using the Pro Multiband Dynamics plug‑in. Though the possibilities presented by using types of compression on different bands — for example an LA‑2A on the bass frequencies and an 1176 on the top end — might be interesting, they also sound like a potential nightmare and I’m happy to stick with a dedicated multiband compressor for these applications. If you feel brave then

Combining Pro Multiband Splitter with a chorus effect applied only to the upper part of the spectrum yields an effect much loved by ‘80s bassists.

help yourself, but I’ll stick with conventional wisdom. The first non-compression example that comes to mind when I think of multiband processing is the bass chorus pedals which I thought were so cool in the ‘80s, but which subsequently left me wondering what I could have been thinking in the ‘90s. The setup involves a two-way split from the bass track with a chorus (or a flanger if you want to go extra-gothic) on the high band only. With the crossover frequency set correctly this keeps the modulation away from the fundamental but can add richness and width to the top end — very ‘80s, just add a fretless and you’re there! You can achieve similar results by high‑pass filtering a duplicate bass part and modulating that, but this seems a better way of doing it, since being able to mute and solo the splits independently makes setting the correct crossover frequency much easier.

Beefy Kicks A subtler application is to use a split to saturate the bottom end of kick drums, basses or anything which has significant deep bass. Any plug-in which distorts can be used, and one of the fascinating things about distortion is how different the characters of different plug-ins can be on the same source. Experimentation is definitely recommended here, but of the stock Pro Tools plug-ins, Avid’s Lo-Fi is an obvious contender. Leave the Sample controls alone and experiment with the Grunge section at the bottom. Distort and Saturation have very different personalities so try both. On the kick drum I was experimenting with I found the best results using Softube’s free Saturation Knob plug-in. These effects can be used in two different ways. Adding harmonics to put some ‘fur’ on the deep bass frequencies, extending them up into the more audible part of the spectrum, is particularly useful if playback on limited‑bandwidth systems

is anticipated. And because distortion becomes progressively more pronounced with level, you can also use this technique to emphasise bass transients.

Wide Open Multiband treatments using time‑ or gain‑based processing are ripe for exploring, but the most consistently useful application I’ve found is in something far more basic: multiband panning. Many of us have seen ‘mono-izer’-style plug-ins, which vary the width of stereo signals with frequency. For good reason these are almost invariably pushing the top end wider while narrowing the bass frequencies down to mono. This can be achieved quite simply in Pro Tools, with Multiband Splitter offering a convenient way to experiment with the width of sounds or even entire mixes. Here’s how to do it: • C  reate a stereo Aux Input and insert Pro Multiband Splitter on it. • Create four more Aux Inputs to route the splits out to. The split outputs from the Splitter can be found under Plug-in, below Input and Bus in the plug-in menu. • Name these four Auxes Low, Low Mid, Mid High and High, and select them all. • Holding Command+Option+Shift, select the Low output from the splitter and the inputs will cascade across all four auxes in the correct order. (If you’re on Windows, hold Control+Alt+Shift). Solo safe these Auxes by Command‑clicking on the solo buttons (Control‑click on a PC) and you’ll be set up.

You should now have five Aux Input tracks, and by routing stereo audio into Splitter it will output through the four split paths. You can adjust the stereo width of different parts of the spectrum using the channel pan controls. To make this easier, click the tiny fader icon on the right‑hand side of the output selector in the Mix window, or in the I/O section of the track headers of the Edit window. This will open the floating fader window, and if you tick both the Link and the Inverse Pan buttons (immediately to the right of the Safe button), then as you turn one pan control the other moves in the opposite direction, making it easy to experiment with adjusting the width of each band. Having done this I’d recommend saving these five tracks as a track preset so they can be recalled in a single operation with the routing intact. Once this is done, bus your desired signals through this setup and experiment. One interesting area worth investigating is using automated multiband panning to add motion to sustained sounds like pads and atmospheres. There is a lot more to multiband splits than esoteric processing choices, so try out these ideas and see how you get on.

By ticking the Link and Inverse Pan buttons, you can easily experiment with the width of each of your separate frequency bands.

w w w . s o u n d o n s o u n d . c o m / September 2020

155

TECHNIQUE / CUBASE

JOHN WALDEN

W

hile archiving your projects is perhaps not the most exciting aspect of working with Cubase, you’ll one day thank your younger self for having put in the effort, so in this month’s column I’ll examine the archival options for users of Cubase Elements, Artist or Pro. Archiving isn’t just about saving your project to another hard drive; it should ensure that you have a backup copy should the original project be lost or become corrupted. It should also ensure that you can access your project even without the current versions of Cubase or the plug-ins you used. The first step is to decide on the balance between your desired degree of future-proofing and the time and effort required to achieve it. I tend towards pragmatism rather than perfectionism, and will focus on the former here. But hopefully there will be enough ideas here that you can develop a more comprehensive strategy if you prefer.

All Backed Up A sensible first step is to create a self-contained backup copy of a project that would restore the project on your own host system in the event of data loss. It’s a two-step process and can be done in an identical fashion in Pro, Artist and Elements.

156

Export Duty Future-proof your projects in Cubase. First, execute the Media/Prepare Archive command. This will check whether all files referenced by your project reside in the project folder, and will place copies of them in there if not. Second, from the menu, select File/Back Up Project, and you’ll be prompted to specify a new folder for the backup copy. A few tickbox options dictate exactly what gets placed in the new folder. Minimise Audio Files (only copy those portions of any audio files actually used on the timeline) and Remove Unused Files will help keep your backup compact. Provided everything else (plug-ins, sample content locations) remains the same on your host system, this backup project should open fully intact and ready to go if your original working copy is lost.

The Future Is Now While useful in the short term, the above process won’t protect you from plug-ins that might go missing in the longer term. The most straightforward insurance against

September 2020 / w w w . s o u n d o n s o u n d . c o m

this problem is to render all the project’s audio and virtual instruments tracks as audio files, with all channel-level processing included. If you like, you can do the same with all the channel processing bypassed, so you have a version of your audio files with any edits, for example, but free from processing, which is handy if you want to revamp your project in the future. What follows will work in all versions of Cubase. It’s a little clunky, though, and users of Pro have a better option in the Export Audio Mixdown dialogue, on which more later. First, save a copy of your project with a suitable ‘version for future-proofing’ name. If the Freeze button isn’t already visible, right-click on any track in the Track List to open the Track Controls Panel and make it visible for both Audio If you are not using Cubase Pro, a workaround combining the track Freeze and Audio Mixdown dialogue features can let you create an audio-only render of all audio and virtual instrument tracks within a project.

and MIDI tracks. Track freezing is actually designed for reducing CPU loads, but it also happens to create an audio render of the frozen track, and places this in a new folder (sensibly named Freeze) in your main Project folder. The render file ‘captures’ both the audio and the results of any channel processing, but there is a catch. In order to keep the file sizes compact, the Freeze function only captures those sections of your track where something is actually happening, and without a little further intervention before freezing, the format of these renders doesn’t really meet our ‘archiving’ needs. For audio tracks, the workaround is simple. First, find a short (a single bar or beat) section of one of your audio tracks that contains silence. Second, copy this clip to the very start of every audio track that doesn’t already start at bar 1, beat 1. Third, select all the audio events on a track (or tracks; you can handle multiple tracks simultaneously) and from the file menu execute Audio/Bounce Selection. This replaces all the separate audio events on each track with a single contiguous event starting at bar 1, beat 1. Finally, if you now Freeze each audio track, the Freeze files will all start from bar 1, beat 1 and can be dragged into a new project. For virtual instrument tracks, a different workaround is required, and must be done one track at a time. In the MixConsole, make sure the output of every virtual

Pro’s Audio Mixdown includes batch processing, making it easier to create a render of all the audio and virtual instrument tracks within a project.

instrument track is set to the main Stereo Output. Next, bypass any plug-ins on the Stereo Output bus (so you don’t end up capturing a ‘mastered’ version of the instrument’s audio). Then set the left locator to bar 1, beat 1 and the right locator to a point just beyond the end of your project. Finally, one VSTi track at a time, solo the track and execute the File/Export/Audio Mixdown command. In the Audio Mixdown dialogue box, specify a filename to reflect the track’s contents and point the File Path to the Freeze folder (which, assuming you’ve already dealt with your audio tracks as described above, will already be there). In the After Export section, deselect the Create Audio Track and Insert To Pool options (so you don’t get extra tracks or audio file entries cluttering up your project) and hit Export Audio. Repeat as required for each virtual instrument track. Then there’s the MIDI: the audio might give you all you need, but you may want to consider archiving the MIDI information used to trigger any virtual or hardware instruments so that you can, for example, choose different voices or layer different parts and so forth at a later date. To do this go to File/Export/MIDI File. You’ll find a few options here, so you should feel free to explore them, but the defaults should suffice.

Go Pro? Cubase Pro makes this all much easier as it allows the simultaneous export of multiple tracks (any track type that uses Using the Audio Mixdown dialogue (shown here in Cubase Elements) allows you to render your virtual instrument tracks to audio, albeit one at a time.

audio) over a time range defined by the left and right locators, so every audio file created starts at the same time point and is exactly the same length. You can define a dedicated folder for the exported files and a naming scheme that can number and name the tracks. (Note that I’ve also deselected the Create Audio Track and Insert To Pool options.) Perhaps the only catch is that you have to choose stereo (the default) or mono bounces; you can’t do both at once. So either choose to live with stereo (no real hardship, unless you’re trying to create a particularly compact archive) or perform the export in a couple of passes, dealing with stereo and mono (tick the Mono Downmix box) separately. Either way, Pro makes this process incredibly efficient, and for users of other Cubase versions, this might be reason enough to upgrade.

Peace Of Mind Yes, it’s fiddly, but if you’ve followed all these steps, you’ll have access not only to a backup of your project, but also a version of the project’s tracks where you can drag and drop all the files to bar 1, beat 1 of a new project. You’ll have a pretty good working version of the original audio, the processed audio, the MIDI information — including the project tempo — and to virtual instrument tracks. Together, these files should meet pretty much any need when it comes to resurrecting a project, be it tomorrow or many years down the line. Whichever route you took, it’s now time to copy your archive to an external drive, and ideally somewhere off-site too. Next time, I’ll expand on these ideas and look at additional export options to help ensure a smooth workflow when collaborating with other musicians or mix engineers. Until then, keep your data safe.

w w w . s o u n d o n s o u n d . c o m / September 2020

157

Q Q

What is a speaker’s crossover frequency?

What exactly is the crossover frequency in studio monitors, and how much attention do I need to pay to that spec? Jem Goodman via email

SOS Technical Editor Hugh Robjohns replies: Most monitor loudspeakers use two (or more) separate loudspeaker drivers, because it’s almost impossible to make a single driver that can cover the full 10-octave audio frequency range with the kind of accuracy and power-handling that is typically required. So it’s common for a monitor loudspeaker to have at least two drivers: a ‘woofer’ or ‘bass-mid’ driver, which handles the lower frequencies, and a ‘tweeter’ to handle the higher frequencies. This would be called a ‘two-way’ monitor speaker, but there are also designs with three drivers (three-way) and more. Whatever the configuration, it’s usually necessary to send only the appropriate frequencies to each driver, and so a ‘crossover’ is required. Essentially, this is a set of complementary high- and low-pass filters. For a two-way design, one filter (a high-pass) allows only the higher frequencies through to the tweeter, and another (low-pass) routes only the low frequencies to the woofer.

Q&A

YOUR TECHNICAL QUESTIONS ANSWERED The ‘crossover frequency’ is where these two filters overlap (see diagram). Typically it will be somewhere between 1 and 3 kHz, but the actual crossover frequency is broadly dependent on the physical size of the tweeter — that determines its power handling and the lowest frequencies it can reproduce — although many other design parameters are taken into consideration. In a two-way system the crossover point often ends up right in the middle of the critical vocal frequency range, and since both drivers are inherently reproducing the signal through the crossover region this can result in some unwanted artifacts, especially if listening off axis. A three-way speaker system avoids this problem as the two crossover regions can be set below and above the critical vocal range.

Q

What cable do I need to connect an RME ADI-4DD to a TC Electronic Reverb 6000?

I’m trying to get an old TC Reverb 6000 MkI connected digitally to my new UAD-Apollo x8 interface to give me four stereo reverb engines. However, the TC6000 has a 25-pin D-sub AES3 connector, which I believe is wired in Yamaha format while the Apollo x8 has

A speaker’s crossover filter determines what frequencies are sent to which driver.

158

September 2020 / w w w . s o u n d o n s o u n d . c o m

Alva make a 0.5m Yamaha-to-Tascam format D-sub cable, but it needs to be connected to a D-sub snake to plug in to the RME ADI-DD.

dual ADAT Toslink connectors, which I understand can support eight channels up to 96k if used together. I intended to use an RME ADI-4DD AES-to-ADAT digital format convertor box, which I remember you have previously reviewed. The RME’s 25-pin D-sub AES3 connector conveniently has an internal jumper which can be changed to swap between different formats, including Yamaha. So, if I change the internal jumper on the RME to the Yamaha D-sub format, what sort of cable do I need? Adrian Mathie via email

SOS Technical Editor Hugh Robjohns replies: Yes, the original TC Electronic Reverb 6000 did indeed use the Yamaha (YGDAI) format for its eight-channel AES3 connector, and the RME ADI-4DD would make an ideal format converter for this situation; I use one myself for this purpose too. However, to use the RME’s D-sub port in the Yamaha format, you’d require a D-sub-to-D-sub cable wired to the Yamaha format at both ends, and I can’t find any commercially available off-the-shelf options any more. Of course, you could have one custom made, but if you were going to do that it would probably make more sense to have it wired as a converter cable (Yamaha-to-AES59/Tascam format) anyway, and leave the RME box in its default AES59/Tascam setting. There are plenty of companies online who could build a suitable custom cable like this. If you’d rather buy off-the-shelf standard cables instead, I’d support RME’s recommendation of using Alva’s digital cables. Alva make a short (0.5m) Yamaha-to-Tascam converter cable which is used as an extension to one of their standard AES59/Tascam-format D-sub‑to‑D-sub cables (available in 1, 3

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or 5 m lengths). I use Alva cables a lot (including Yamaha converter cables) and find them to be both reliable and very good value. This one has the male Yamaha connector, which you can plug into the Reverb 6000, but a female Tascam connector at the other end. You need a male Tascam connector to plug into the ADI-4DD’s D-sub. If the units are close by, a simple D-sub male-to-male gender changer would be the least expensive way to do that, but it might be a bit cumbersome hanging off the back of the RME. Probably a better option then is a regular D-sub-to-D-sub cable. You can then run Toslink optical cables to the Apollo interface. In an extended digital system like this clocking is important. Although you could use the Toslink connectors to provide clocks, I’d recommend using a BNC cable from the Apollo’s word-clock output to the RME’s clock in, configuring the Apollo as the lead clock and the RME as a follower. The Reverb 6000 should be configured to follow the clock of its AES inputs.

Q

What’s the best way to clean dust from my gear?

My home studio is plagued by dust! It seems to get everywhere. Obviously, I vacuum and so on, but it’s really hard to remove from my keyboards, my mixer and the various cables that hook it all up — I keep sucking up little things like screws and plectrums. Is there a ‘right’ way to clean this stuff? Dan Hill via email

SOS Reviews Editor Matt Houghton replies: I’m not sure about a ‘right’ way, but there are certainly ways and means. The simplest way I find to tackle dust on fiddly surfaces with knobs, keys or faders on them is to use a vacuum cleaner, but to use a paint brush or old-fashioned shaving brush to loosen the dust from all the nooks and crannies; as it comes off, it flies straight into the vacuum cleaner. Most of the cheapo handheld things don’t seem to have enough ‘suck’, and while a good chunk of money can buy you a more powerful one, I prefer to use a regular vacuum cleaner with the head detached, and stop it from guzzling tiny gadgets by putting a filter over the end of the tube that allows air and dust to pass through but nothing larger. If anyone

160

Protecting your gear from dust needn’t cost a fortune!

in your house is throwing out some old tights, that material will do nicely: attach firmly with a rubber band and you can de-dust to your heart’s content. Don’t forget to clean the dust from all the less obvious places it tends to collect, like the tops of any acoustic panels, shelves, door frames, keyboard stands, chair legs, underneath sofas and so on. I find that cables can be real dust magnets, especially if you have an old-school analogue setup with a console, patchbay and multicore snakes everywhere. A damp cloth will get rid of most of it easily enough, so it’s worth making sure you can easily get behind any racks or console. Once you’ve given the place a good deep clean, the surest way to keep the dust at bay from something like a mixer or keyboard is to invest in dust covers. Dedicated ones come in various forms, from branded accessories to match your keyboard, to universal stretchable spandex ones. If you’re a cheapskate like me, unfitted cotton twill dust sheets from the DIY store work well. Not only can you cut them up to cover several bits of gear with one sheet, but you can chuck them in the washing machine from time to time.

September 2020 / w w w . s o u n d o n s o u n d . c o m

It’s worth noting that this isn’t just an aesthetic issue: a lot of mixers have jack sockets on the top panel, and dust falling into these can cause problems in the signal path (interruptions, crackles and so forth). The dust cover will also protect against that too. If it’s already a problem, try squirting in a bit of Deoxit D5 and pushing a lightly burred (ie. sandpapered) jack plug in and out of the socket. Another thing to consider is oil and grease. Every time you put your mucky fingers on your gear, you leave behind a thin layer of grease, not to mention the residue of any sneaky studio snacks you may have been indulging in. You might not even see it, but the grease can be a bit of a dust/dirt magnet. I usually use isopropyl alcohol to remove it but you could also try glass cleaner. It’s better to put this directly on the cloth and rub the surface than spray the liquid on your gear. Needless to say, don’t use a cloth that will leave fibres behind; microfibre ones such as those used to clean spectacles and camera lenses seem to work pretty well. That’s about it for the day-to-day stuff — cleaning inside faders and pots is a whole different story!

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WHY I LOVE...

THE JBL CONTROL ONE HELEN SKIERA

W

hen asked “What is your favourite loudspeaker?” I feel, as a sound designer in theatre and live performance, the answer ought to reflect the stunning selection of d&B, Meyer and E&M gloriousness that I get to experience on a regular basis. Publicly it will be but, secretly, my love is reserved for a tiny little box that has been my saviour in many situations. For theatre designers, the first shows of our career are often low/no‑pay situations in low/no‑tech function spaces. In one particular venue, a room above a pub, the ‘sound system’ (I use this term in its loosest sense) did emit some sort of audio, but any serious resemblance to loudspeakers ended there. It was not the unique qualities of the speakers that were initially the most notable, but their positioning: one was on the floor, and the other on top of a wardrobe.

Why there was a wardrobe at all in this theatre I could not explain then or now, but I did ask the appointed sound technical person if it would be possible to have them at the same height. This particular technician was, as so many were in those days, male, bearded (in an ‘I haven’t seen soap or a razor since the ‘70s’ way, rather than hipster cool), deeply unfriendly, and with all the sartorial elegance of Stig of the Dump. He fixed me with a gaze and said, “No one’s ever complained before.” With the rigging choices of ‘on top of a wardrobe’ or ‘not on top of a wardrobe’, I opted for the latter. I lifted down the offending unit and discovered that the two boxes were entirely different flavours of thing. Audio output sounded like it was passing through particularly tired marshmallows — two different types of marshmallow, of course. The very lovely Royal Court Theatre, where I had a real job at the same time as designing, loaned me four JBL Control

Ones. These being so light, I could rig them from the overhead grid, and this gave surprisingly good coverage in the audience area. But how good is the sound from a Control One? Well, compared to hearing audio through two different wardrobe-mounted (or not) fluffy-floofy, clarity-free boxes of doom, really very good indeed. I remember listening to a Handel extract that I had carefully carved out using Audacity for a particular cue, and at that moment, it was one of the best things I had ever heard. Context counts for a lot, and is perhaps worth remembering in every aspect of creating audio for a live performance. As the shows got bigger, the rigs got bigger and better, but there is still room in the biggest of rigs for a Control One. A great thing about them is that they are so inexpensive, meaning that if something bad happens to them they can be replaced.

NEXT MONTH IN Jarvis Cocker We talk to living legend Jarvis Cocker about his latest JARV IS... album, and the unique approaches of the many producers he’s worked with throughout his extraordinary career.

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“So why don’t you just look after your speakers?” I hear you cry. Well, in theatre we sometimes need sound to come out of the ground, or through a wall, or be in the same place that someone spills water every night. Control Ones with dented grilles, sticky with fake blood or painted bright yellow for some forgotten scenery reason are the unsung heroes of theatre audio. Front of house, the main system takes all the glory, but right in the action, on‑stage speakers are often vital. If the doorbell or a phone doesn’t ring when it should, your show will be severely affected, and the audio needs to be situated in a realistic space that might be subject to all sorts of abuse. So when we’re told, “The designer needs something to make the sound of frying oil, then have a bag of gravel emptied over it, and in the next scene they pour petrol over it and set fire to the whole thing,” the only answer can be, “Let’s use a Control One.”  

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